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https://github.com/FFmpeg/FFmpeg.git
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avfilter: add adrc filter
This commit is contained in:
Paul B Mahol
parent
744100af62
commit
3b66757d7d
@ -27,6 +27,7 @@ version <next>:
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- QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
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- showcwt multimedia filter
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- corr video filter
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- adrc audio filter
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version 5.1:
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@ -872,6 +872,91 @@ Compute derivative/integral of audio stream.
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Applying both filters one after another produces original audio.
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@section adrc
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Apply spectral dynamic range controller filter to input audio stream.
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A description of the accepted options follows.
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@table @option
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@item transfer
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Set the transfer expression.
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The expression can contain the following constants:
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@table @option
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@item ch
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current channel number
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@item sn
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current sample number
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@item nb_channels
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number of channels
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@item t
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timestamp expressed in seconds
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@item sr
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sample rate
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@item p
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current frequency power value, in dB
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@item f
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current frequency in Hz
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@end table
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Default value is @code{p}.
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@item attack
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Set the attack in milliseconds. Default is @code{50} milliseconds.
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Allowed range is from 1 to 1000 milliseconds.
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@item release
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Set the release in milliseconds. Default is @code{100} milliseconds.
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Allowed range is from 5 to 2000 milliseconds.
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@item channels
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Set which channels to filter, by default @code{all} channels in audio stream are filtered.
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@end table
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@subsection Commands
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This filter supports the all above options as @ref{commands}.
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@subsection Examples
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@itemize
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@item
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Apply spectral compression to all frequencies with threshold of -50 dB and 1:6 ratio:
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@example
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adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100
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@end example
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@item
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Similar to above but with 1:2 ratio and filtering only front center channel:
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@example
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adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC
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@end example
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@item
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Apply spectral noise gate to all frequencies with threshold of -85 dB and with short attack time and short release time:
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@example
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adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5
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@end example
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@item
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Apply spectral expansion to all frequencies with threshold of -10 dB and 1:2 ratio:
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@example
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adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100
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@end example
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@item
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Apply limiter to max -60 dB to all frequencies, with attack of 2 ms and release of 10 ms:
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@example
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adrc=transfer='min(p,-60)':attack=2:release=10
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@end example
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@end itemize
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@section adynamicequalizer
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Apply dynamic equalization to input audio stream.
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@ -48,6 +48,7 @@ OBJS-$(CONFIG_ADECORRELATE_FILTER) += af_adecorrelate.o
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OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
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OBJS-$(CONFIG_ADENORM_FILTER) += af_adenorm.o
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OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
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OBJS-$(CONFIG_ADRC_FILTER) += af_adrc.o
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OBJS-$(CONFIG_ADYNAMICEQUALIZER_FILTER) += af_adynamicequalizer.o
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OBJS-$(CONFIG_ADYNAMICSMOOTH_FILTER) += af_adynamicsmooth.o
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OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
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508
libavfilter/af_adrc.c
Normal file
508
libavfilter/af_adrc.c
Normal file
@ -0,0 +1,508 @@
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/*
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* Copyright (c) 2022 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include "libavutil/eval.h"
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#include "libavutil/ffmath.h"
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#include "libavutil/opt.h"
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#include "libavutil/tx.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "internal.h"
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static const char * const var_names[] = {
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"ch", ///< the value of the current channel
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"sn", ///< number of samples
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"nb_channels",
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"t", ///< timestamp expressed in seconds
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"sr", ///< sample rate
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"p", ///< input power in dB for frequency bin
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"f", ///< frequency in Hz
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NULL
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};
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enum var_name {
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VAR_CH,
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VAR_SN,
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VAR_NB_CHANNELS,
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VAR_T,
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VAR_SR,
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VAR_P,
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VAR_F,
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VAR_VARS_NB
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};
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typedef struct AudioDRCContext {
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const AVClass *class;
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double attack_ms;
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double release_ms;
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char *expr_str;
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double attack;
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double release;
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int fft_size;
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int overlap;
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int channels;
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float fx;
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float *window;
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AVFrame *drc_frame;
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AVFrame *energy;
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AVFrame *envelope;
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AVFrame *factors;
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AVFrame *in;
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AVFrame *in_buffer;
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AVFrame *in_frame;
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AVFrame *out_dist_frame;
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AVFrame *spectrum_buf;
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AVFrame *target_gain;
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AVFrame *windowed_frame;
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char *channels_to_filter;
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AVChannelLayout ch_layout;
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AVTXContext **tx_ctx;
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av_tx_fn tx_fn;
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AVTXContext **itx_ctx;
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av_tx_fn itx_fn;
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AVExpr *expr;
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double var_values[VAR_VARS_NB];
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} AudioDRCContext;
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#define OFFSET(x) offsetof(AudioDRCContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption adrc_options[] = {
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{ "transfer", "set the transfer expression", OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str="p"}, 0, 0, FLAGS },
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{ "attack", "set the attack", OFFSET(attack_ms), AV_OPT_TYPE_DOUBLE, {.dbl=50.}, 1, 1000, FLAGS },
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{ "release", "set the release", OFFSET(release_ms), AV_OPT_TYPE_DOUBLE, {.dbl=100.}, 5, 2000, FLAGS },
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{ "channels", "set channels to filter",OFFSET(channels_to_filter),AV_OPT_TYPE_STRING,{.str="all"},0, 0, FLAGS },
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{NULL}
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};
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AVFILTER_DEFINE_CLASS(adrc);
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static void generate_hann_window(float *window, int size)
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{
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for (int i = 0; i < size; i++) {
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float value = 0.5f * (1.f - cosf(2.f * M_PI * i / size));
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window[i] = value;
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}
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDRCContext *s = ctx->priv;
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float scale;
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int ret;
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s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
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s->fx = inlink->sample_rate * 0.5f / (s->fft_size / 2 + 1);
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s->overlap = s->fft_size / 4;
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s->window = av_calloc(s->fft_size, sizeof(*s->window));
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if (!s->window)
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return AVERROR(ENOMEM);
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s->drc_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->energy = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
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s->envelope = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
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s->factors = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
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s->in_buffer = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->target_gain = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
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s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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if (!s->in_buffer || !s->in_frame || !s->target_gain ||
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!s->out_dist_frame || !s->windowed_frame || !s->envelope ||
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!s->drc_frame || !s->spectrum_buf || !s->energy || !s->factors)
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return AVERROR(ENOMEM);
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generate_hann_window(s->window, s->fft_size);
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s->channels = inlink->ch_layout.nb_channels;
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s->tx_ctx = av_calloc(s->channels, sizeof(*s->tx_ctx));
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s->itx_ctx = av_calloc(s->channels, sizeof(*s->itx_ctx));
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if (!s->tx_ctx || !s->itx_ctx)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < s->channels; ch++) {
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scale = 1.f / s->fft_size;
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ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_RDFT, 0, s->fft_size, &scale, 0);
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if (ret < 0)
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return ret;
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scale = 1.f;
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ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_RDFT, 1, s->fft_size, &scale, 0);
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if (ret < 0)
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return ret;
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}
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s->var_values[VAR_SR] = inlink->sample_rate;
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s->var_values[VAR_NB_CHANNELS] = s->channels;
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return av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
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NULL, NULL, 0, ctx);
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}
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static void apply_window(AudioDRCContext *s,
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const float *in_frame, float *out_frame, const int add_to_out_frame)
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{
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const float *window = s->window;
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const int fft_size = s->fft_size;
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if (add_to_out_frame) {
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for (int i = 0; i < fft_size; i++)
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out_frame[i] += in_frame[i] * window[i];
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} else {
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for (int i = 0; i < fft_size; i++)
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out_frame[i] = in_frame[i] * window[i];
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}
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}
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static float sqrf(float x)
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{
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return x * x;
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}
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static void get_energy(AVFilterContext *ctx,
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int len,
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float *energy,
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const float *spectral)
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{
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for (int n = 0; n < len; n++) {
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energy[n] = 10.f * log10f(sqrf(spectral[2 * n]) + sqrf(spectral[2 * n + 1]));
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if (!isnormal(energy[n]))
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energy[n] = -351.f;
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}
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}
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static void get_target_gain(AVFilterContext *ctx,
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int len,
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float *gain,
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const float *energy,
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double *var_values,
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float fx, int bypass)
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{
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AudioDRCContext *s = ctx->priv;
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if (bypass) {
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memcpy(gain, energy, sizeof(*gain) * len);
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return;
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}
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for (int n = 0; n < len; n++) {
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const float Xg = energy[n];
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var_values[VAR_P] = Xg;
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var_values[VAR_F] = n * fx;
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gain[n] = av_expr_eval(s->expr, var_values, s);
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}
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}
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static void get_envelope(AVFilterContext *ctx,
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int len,
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float *envelope,
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const float *energy,
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const float *gain)
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{
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AudioDRCContext *s = ctx->priv;
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const float release = s->release;
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const float attack = s->attack;
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for (int n = 0; n < len; n++) {
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const float Bg = gain[n] - energy[n];
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const float Vg = envelope[n];
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if (Bg > Vg) {
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envelope[n] = attack * Vg + (1.f - attack) * Bg;
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} else if (Bg <= Vg) {
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envelope[n] = release * Vg + (1.f - release) * Bg;
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} else {
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envelope[n] = 0.f;
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}
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}
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}
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static void get_factors(AVFilterContext *ctx,
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int len,
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float *factors,
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const float *envelope)
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{
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for (int n = 0; n < len; n++)
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factors[n] = sqrtf(ff_exp10f(envelope[n] / 10.f));
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}
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static void apply_factors(AVFilterContext *ctx,
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int len,
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float *spectrum,
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const float *factors)
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{
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for (int n = 0; n < len; n++) {
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spectrum[2*n+0] *= factors[n];
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spectrum[2*n+1] *= factors[n];
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}
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}
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static void feed(AVFilterContext *ctx, int ch,
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const float *in_samples, float *out_samples,
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float *in_frame, float *out_dist_frame,
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float *windowed_frame, float *drc_frame,
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float *spectrum_buf, float *energy,
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float *target_gain, float *envelope,
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float *factors)
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{
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AudioDRCContext *s = ctx->priv;
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double var_values[VAR_VARS_NB];
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const int fft_size = s->fft_size;
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const int nb_coeffs = s->fft_size / 2 + 1;
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const int overlap = s->overlap;
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enum AVChannel channel = av_channel_layout_channel_from_index(&ctx->inputs[0]->ch_layout, ch);
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const int bypass = av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0;
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memcpy(var_values, s->var_values, sizeof(var_values));
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var_values[VAR_CH] = ch;
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// shift in/out buffers
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memmove(in_frame, in_frame + overlap, (fft_size - overlap) * sizeof(*in_frame));
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memmove(out_dist_frame, out_dist_frame + overlap, (fft_size - overlap) * sizeof(*out_dist_frame));
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memcpy(in_frame + fft_size - overlap, in_samples, sizeof(*in_frame) * overlap);
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memset(out_dist_frame + fft_size - overlap, 0, sizeof(*out_dist_frame) * overlap);
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apply_window(s, in_frame, windowed_frame, 0);
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s->tx_fn(s->tx_ctx[ch], spectrum_buf, windowed_frame, sizeof(float));
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get_energy(ctx, nb_coeffs, energy, spectrum_buf);
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get_target_gain(ctx, nb_coeffs, target_gain, energy, var_values, s->fx, bypass);
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get_envelope(ctx, nb_coeffs, envelope, energy, target_gain);
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get_factors(ctx, nb_coeffs, factors, envelope);
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apply_factors(ctx, nb_coeffs, spectrum_buf, factors);
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s->itx_fn(s->itx_ctx[ch], drc_frame, spectrum_buf, sizeof(AVComplexFloat));
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apply_window(s, drc_frame, out_dist_frame, 1);
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// 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
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if (!ctx->is_disabled) {
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for (int i = 0; i < overlap; i++)
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out_samples[i] = out_dist_frame[i] / 1.5f;
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} else {
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memcpy(out_samples, in_frame, sizeof(*out_samples) * overlap);
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}
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}
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static int drc_channel(AVFilterContext *ctx, AVFrame *in, AVFrame *out, int ch)
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{
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AudioDRCContext *s = ctx->priv;
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const float *src = (const float *)in->extended_data[ch];
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float *in_buffer = (float *)s->in_buffer->extended_data[ch];
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float *dst = (float *)out->extended_data[ch];
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memcpy(in_buffer, src, sizeof(*in_buffer) * s->overlap);
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feed(ctx, ch, in_buffer, dst,
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(float *)(s->in_frame->extended_data[ch]),
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(float *)(s->out_dist_frame->extended_data[ch]),
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(float *)(s->windowed_frame->extended_data[ch]),
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(float *)(s->drc_frame->extended_data[ch]),
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(float *)(s->spectrum_buf->extended_data[ch]),
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(float *)(s->energy->extended_data[ch]),
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(float *)(s->target_gain->extended_data[ch]),
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(float *)(s->envelope->extended_data[ch]),
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(float *)(s->factors->extended_data[ch]));
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return 0;
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}
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|
||||
static int drc_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
|
||||
{
|
||||
AudioDRCContext *s = ctx->priv;
|
||||
AVFrame *in = s->in;
|
||||
AVFrame *out = arg;
|
||||
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
|
||||
const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
|
||||
|
||||
for (int ch = start; ch < end; ch++)
|
||||
drc_channel(ctx, in, out, ch);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
||||
{
|
||||
AVFilterContext *ctx = inlink->dst;
|
||||
AVFilterLink *outlink = ctx->outputs[0];
|
||||
AudioDRCContext *s = ctx->priv;
|
||||
AVFrame *out;
|
||||
int ret;
|
||||
|
||||
out = ff_get_audio_buffer(outlink, s->overlap);
|
||||
if (!out) {
|
||||
ret = AVERROR(ENOMEM);
|
||||
goto fail;
|
||||
}
|
||||
|
||||
s->var_values[VAR_SN] = outlink->sample_count_in;
|
||||
s->var_values[VAR_T] = s->var_values[VAR_SN] * (double)1/outlink->sample_rate;
|
||||
|
||||
s->in = in;
|
||||
av_frame_copy_props(out, in);
|
||||
ff_filter_execute(ctx, drc_channels, out, NULL,
|
||||
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
|
||||
|
||||
out->pts = in->pts;
|
||||
out->nb_samples = in->nb_samples;
|
||||
ret = ff_filter_frame(outlink, out);
|
||||
fail:
|
||||
av_frame_free(&in);
|
||||
s->in = NULL;
|
||||
return ret < 0 ? ret : 0;
|
||||
}
|
||||
|
||||
static int activate(AVFilterContext *ctx)
|
||||
{
|
||||
AVFilterLink *inlink = ctx->inputs[0];
|
||||
AVFilterLink *outlink = ctx->outputs[0];
|
||||
AudioDRCContext *s = ctx->priv;
|
||||
AVFrame *in = NULL;
|
||||
int ret = 0, status;
|
||||
int64_t pts;
|
||||
|
||||
ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
if (strcmp(s->channels_to_filter, "all"))
|
||||
av_channel_layout_from_string(&s->ch_layout, s->channels_to_filter);
|
||||
|
||||
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
|
||||
|
||||
ret = ff_inlink_consume_samples(inlink, s->overlap, s->overlap, &in);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
|
||||
if (ret > 0) {
|
||||
s->attack = expf(-1.f / (s->attack_ms * inlink->sample_rate / 1000.f));
|
||||
s->release = expf(-1.f / (s->release_ms * inlink->sample_rate / 1000.f));
|
||||
|
||||
return filter_frame(inlink, in);
|
||||
} else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
|
||||
ff_outlink_set_status(outlink, status, pts);
|
||||
return 0;
|
||||
} else {
|
||||
if (ff_inlink_queued_samples(inlink) >= s->overlap) {
|
||||
ff_filter_set_ready(ctx, 10);
|
||||
} else if (ff_outlink_frame_wanted(outlink)) {
|
||||
ff_inlink_request_frame(inlink);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
static av_cold void uninit(AVFilterContext *ctx)
|
||||
{
|
||||
AudioDRCContext *s = ctx->priv;
|
||||
|
||||
av_channel_layout_uninit(&s->ch_layout);
|
||||
|
||||
av_expr_free(s->expr);
|
||||
s->expr = NULL;
|
||||
|
||||
av_freep(&s->window);
|
||||
|
||||
av_frame_free(&s->drc_frame);
|
||||
av_frame_free(&s->energy);
|
||||
av_frame_free(&s->envelope);
|
||||
av_frame_free(&s->factors);
|
||||
av_frame_free(&s->in_buffer);
|
||||
av_frame_free(&s->in_frame);
|
||||
av_frame_free(&s->out_dist_frame);
|
||||
av_frame_free(&s->spectrum_buf);
|
||||
av_frame_free(&s->target_gain);
|
||||
av_frame_free(&s->windowed_frame);
|
||||
|
||||
for (int ch = 0; ch < s->channels; ch++) {
|
||||
if (s->tx_ctx)
|
||||
av_tx_uninit(&s->tx_ctx[ch]);
|
||||
if (s->itx_ctx)
|
||||
av_tx_uninit(&s->itx_ctx[ch]);
|
||||
}
|
||||
|
||||
av_freep(&s->tx_ctx);
|
||||
av_freep(&s->itx_ctx);
|
||||
}
|
||||
|
||||
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
|
||||
char *res, int res_len, int flags)
|
||||
{
|
||||
AudioDRCContext *s = ctx->priv;
|
||||
char *old_expr_str = av_strdup(s->expr_str);
|
||||
int ret;
|
||||
|
||||
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
|
||||
if (ret >= 0 && strcmp(old_expr_str, s->expr_str)) {
|
||||
ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
|
||||
NULL, NULL, 0, ctx);
|
||||
}
|
||||
av_free(old_expr_str);
|
||||
return ret;
|
||||
}
|
||||
|
||||
static const AVFilterPad inputs[] = {
|
||||
{
|
||||
.name = "default",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
.config_props = config_input,
|
||||
},
|
||||
};
|
||||
|
||||
static const AVFilterPad outputs[] = {
|
||||
{
|
||||
.name = "default",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
},
|
||||
};
|
||||
|
||||
const AVFilter ff_af_adrc = {
|
||||
.name = "adrc",
|
||||
.description = NULL_IF_CONFIG_SMALL("Audio Spectral Dynamic Range Controller."),
|
||||
.priv_size = sizeof(AudioDRCContext),
|
||||
.priv_class = &adrc_class,
|
||||
.uninit = uninit,
|
||||
FILTER_INPUTS(inputs),
|
||||
FILTER_OUTPUTS(outputs),
|
||||
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
|
||||
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
|
||||
AVFILTER_FLAG_SLICE_THREADS,
|
||||
.activate = activate,
|
||||
.process_command = process_command,
|
||||
};
|
||||
@ -35,6 +35,7 @@ extern const AVFilter ff_af_adecorrelate;
|
||||
extern const AVFilter ff_af_adelay;
|
||||
extern const AVFilter ff_af_adenorm;
|
||||
extern const AVFilter ff_af_aderivative;
|
||||
extern const AVFilter ff_af_adrc;
|
||||
extern const AVFilter ff_af_adynamicequalizer;
|
||||
extern const AVFilter ff_af_adynamicsmooth;
|
||||
extern const AVFilter ff_af_aecho;
|
||||
|
||||
@ -31,7 +31,7 @@
|
||||
|
||||
#include "version_major.h"
|
||||
|
||||
#define LIBAVFILTER_VERSION_MINOR 52
|
||||
#define LIBAVFILTER_VERSION_MINOR 53
|
||||
#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user