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avfilter: add adrc filter

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pull/388/head
Paul B Mahol 2 years ago
parent 744100af62
commit 3b66757d7d
6 changed files with 597 additions and 1 deletions
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  1. 1
      Changelog
  2. 85
      doc/filters.texi
  3. 1
      libavfilter/Makefile
  4. 508
      libavfilter/af_adrc.c
  5. 1
      libavfilter/allfilters.c
  6. 2
      libavfilter/version.h

1
Changelog
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@ -27,6 +27,7 @@ version <next>:
- QSV decoding and encoding for 10/12bit 422, 10/12bit 444 HEVC and VP9
- showcwt multimedia filter
- corr video filter
- adrc audio filter
version 5.1:

85
doc/filters.texi
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@ -872,6 +872,91 @@ Compute derivative/integral of audio stream.
Applying both filters one after another produces original audio.
@section adrc
Apply spectral dynamic range controller filter to input audio stream.
A description of the accepted options follows.
@table @option
@item transfer
Set the transfer expression.
The expression can contain the following constants:
@table @option
@item ch
current channel number
@item sn
current sample number
@item nb_channels
number of channels
@item t
timestamp expressed in seconds
@item sr
sample rate
@item p
current frequency power value, in dB
@item f
current frequency in Hz
@end table
Default value is @code{p}.
@item attack
Set the attack in milliseconds. Default is @code{50} milliseconds.
Allowed range is from 1 to 1000 milliseconds.
@item release
Set the release in milliseconds. Default is @code{100} milliseconds.
Allowed range is from 5 to 2000 milliseconds.
@item channels
Set which channels to filter, by default @code{all} channels in audio stream are filtered.
@end table
@subsection Commands
This filter supports the all above options as @ref{commands}.
@subsection Examples
@itemize
@item
Apply spectral compression to all frequencies with threshold of -50 dB and 1:6 ratio:
@example
adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100
@end example
@item
Similar to above but with 1:2 ratio and filtering only front center channel:
@example
adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC
@end example
@item
Apply spectral noise gate to all frequencies with threshold of -85 dB and with short attack time and short release time:
@example
adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5
@end example
@item
Apply spectral expansion to all frequencies with threshold of -10 dB and 1:2 ratio:
@example
adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100
@end example
@item
Apply limiter to max -60 dB to all frequencies, with attack of 2 ms and release of 10 ms:
@example
adrc=transfer='min(p,-60)':attack=2:release=10
@end example
@end itemize
@section adynamicequalizer
Apply dynamic equalization to input audio stream.

1
libavfilter/Makefile
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@ -48,6 +48,7 @@ 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
OBJS-$(CONFIG_ADRC_FILTER) += af_adrc.o
OBJS-$(CONFIG_ADYNAMICEQUALIZER_FILTER) += af_adynamicequalizer.o
OBJS-$(CONFIG_ADYNAMICSMOOTH_FILTER) += af_adynamicsmooth.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o

508
libavfilter/af_adrc.c
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@ -0,0 +1,508 @@
/*
* Copyright (c) 2022 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 <float.h>
#include "libavutil/eval.h"
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
static const char * const var_names[] = {
"ch", ///< the value of the current channel
"sn", ///< number of samples
"nb_channels",
"t", ///< timestamp expressed in seconds
"sr", ///< sample rate
"p", ///< input power in dB for frequency bin
"f", ///< frequency in Hz
NULL
};
enum var_name {
VAR_CH,
VAR_SN,
VAR_NB_CHANNELS,
VAR_T,
VAR_SR,
VAR_P,
VAR_F,
VAR_VARS_NB
};
typedef struct AudioDRCContext {
const AVClass *class;
double attack_ms;
double release_ms;
char *expr_str;
double attack;
double release;
int fft_size;
int overlap;
int channels;
float fx;
float *window;
AVFrame *drc_frame;
AVFrame *energy;
AVFrame *envelope;
AVFrame *factors;
AVFrame *in;
AVFrame *in_buffer;
AVFrame *in_frame;
AVFrame *out_dist_frame;
AVFrame *spectrum_buf;
AVFrame *target_gain;
AVFrame *windowed_frame;
char *channels_to_filter;
AVChannelLayout ch_layout;
AVTXContext **tx_ctx;
av_tx_fn tx_fn;
AVTXContext **itx_ctx;
av_tx_fn itx_fn;
AVExpr *expr;
double var_values[VAR_VARS_NB];
} AudioDRCContext;
#define OFFSET(x) offsetof(AudioDRCContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption adrc_options[] = {
{ "transfer", "set the transfer expression", OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str="p"}, 0, 0, FLAGS },
{ "attack", "set the attack", OFFSET(attack_ms), AV_OPT_TYPE_DOUBLE, {.dbl=50.}, 1, 1000, FLAGS },
{ "release", "set the release", OFFSET(release_ms), AV_OPT_TYPE_DOUBLE, {.dbl=100.}, 5, 2000, FLAGS },
{ "channels", "set channels to filter",OFFSET(channels_to_filter),AV_OPT_TYPE_STRING,{.str="all"},0, 0, FLAGS },
{NULL}
};
AVFILTER_DEFINE_CLASS(adrc);
static void generate_hann_window(float *window, int size)
{
for (int i = 0; i < size; i++) {
float value = 0.5f * (1.f - cosf(2.f * M_PI * i / size));
window[i] = value;
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioDRCContext *s = ctx->priv;
float scale;
int ret;
s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
s->fx = inlink->sample_rate * 0.5f / (s->fft_size / 2 + 1);
s->overlap = s->fft_size / 4;
s->window = av_calloc(s->fft_size, sizeof(*s->window));
if (!s->window)
return AVERROR(ENOMEM);
s->drc_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->energy = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
s->envelope = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
s->factors = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
s->in_buffer = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->target_gain = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
if (!s->in_buffer || !s->in_frame || !s->target_gain ||
!s->out_dist_frame || !s->windowed_frame || !s->envelope ||
!s->drc_frame || !s->spectrum_buf || !s->energy || !s->factors)
return AVERROR(ENOMEM);
generate_hann_window(s->window, s->fft_size);
s->channels = inlink->ch_layout.nb_channels;
s->tx_ctx = av_calloc(s->channels, sizeof(*s->tx_ctx));
s->itx_ctx = av_calloc(s->channels, sizeof(*s->itx_ctx));
if (!s->tx_ctx || !s->itx_ctx)
return AVERROR(ENOMEM);
for (int ch = 0; ch < s->channels; ch++) {
scale = 1.f / s->fft_size;
ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_RDFT, 0, s->fft_size, &scale, 0);
if (ret < 0)
return ret;
scale = 1.f;
ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_RDFT, 1, s->fft_size, &scale, 0);
if (ret < 0)
return ret;
}
s->var_values[VAR_SR] = inlink->sample_rate;
s->var_values[VAR_NB_CHANNELS] = s->channels;
return av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
NULL, NULL, 0, ctx);
}
static void apply_window(AudioDRCContext *s,
const float *in_frame, float *out_frame, const int add_to_out_frame)
{
const float *window = s->window;
const int fft_size = s->fft_size;
if (add_to_out_frame) {
for (int i = 0; i < fft_size; i++)
out_frame[i] += in_frame[i] * window[i];
} else {
for (int i = 0; i < fft_size; i++)
out_frame[i] = in_frame[i] * window[i];
}
}
static float sqrf(float x)
{
return x * x;
}
static void get_energy(AVFilterContext *ctx,
int len,
float *energy,
const float *spectral)
{
for (int n = 0; n < len; n++) {
energy[n] = 10.f * log10f(sqrf(spectral[2 * n]) + sqrf(spectral[2 * n + 1]));
if (!isnormal(energy[n]))
energy[n] = -351.f;
}
}
static void get_target_gain(AVFilterContext *ctx,
int len,
float *gain,
const float *energy,
double *var_values,
float fx, int bypass)
{
AudioDRCContext *s = ctx->priv;
if (bypass) {
memcpy(gain, energy, sizeof(*gain) * len);
return;
}
for (int n = 0; n < len; n++) {
const float Xg = energy[n];
var_values[VAR_P] = Xg;
var_values[VAR_F] = n * fx;
gain[n] = av_expr_eval(s->expr, var_values, s);
}
}
static void get_envelope(AVFilterContext *ctx,
int len,
float *envelope,
const float *energy,
const float *gain)
{
AudioDRCContext *s = ctx->priv;
const float release = s->release;
const float attack = s->attack;
for (int n = 0; n < len; n++) {
const float Bg = gain[n] - energy[n];
const float Vg = envelope[n];
if (Bg > Vg) {
envelope[n] = attack * Vg + (1.f - attack) * Bg;
} else if (Bg <= Vg) {
envelope[n] = release * Vg + (1.f - release) * Bg;
} else {
envelope[n] = 0.f;
}
}
}
static void get_factors(AVFilterContext *ctx,
int len,
float *factors,
const float *envelope)
{
for (int n = 0; n < len; n++)
factors[n] = sqrtf(ff_exp10f(envelope[n] / 10.f));
}
static void apply_factors(AVFilterContext *ctx,
int len,
float *spectrum,
const float *factors)
{
for (int n = 0; n < len; n++) {
spectrum[2*n+0] *= factors[n];
spectrum[2*n+1] *= factors[n];
}
}
static void feed(AVFilterContext *ctx, int ch,
const float *in_samples, float *out_samples,
float *in_frame, float *out_dist_frame,
float *windowed_frame, float *drc_frame,
float *spectrum_buf, float *energy,
float *target_gain, float *envelope,
float *factors)
{
AudioDRCContext *s = ctx->priv;
double var_values[VAR_VARS_NB];
const int fft_size = s->fft_size;
const int nb_coeffs = s->fft_size / 2 + 1;
const int overlap = s->overlap;
enum AVChannel channel = av_channel_layout_channel_from_index(&ctx->inputs[0]->ch_layout, ch);
const int bypass = av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0;
memcpy(var_values, s->var_values, sizeof(var_values));
var_values[VAR_CH] = ch;
// shift in/out buffers
memmove(in_frame, in_frame + overlap, (fft_size - overlap) * sizeof(*in_frame));
memmove(out_dist_frame, out_dist_frame + overlap, (fft_size - overlap) * sizeof(*out_dist_frame));
memcpy(in_frame + fft_size - overlap, in_samples, sizeof(*in_frame) * overlap);
memset(out_dist_frame + fft_size - overlap, 0, sizeof(*out_dist_frame) * overlap);
apply_window(s, in_frame, windowed_frame, 0);
s->tx_fn(s->tx_ctx[ch], spectrum_buf, windowed_frame, sizeof(float));
get_energy(ctx, nb_coeffs, energy, spectrum_buf);
get_target_gain(ctx, nb_coeffs, target_gain, energy, var_values, s->fx, bypass);
get_envelope(ctx, nb_coeffs, envelope, energy, target_gain);
get_factors(ctx, nb_coeffs, factors, envelope);
apply_factors(ctx, nb_coeffs, spectrum_buf, factors);
s->itx_fn(s->itx_ctx[ch], drc_frame, spectrum_buf, sizeof(AVComplexFloat));
apply_window(s, drc_frame, out_dist_frame, 1);
// 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
if (!ctx->is_disabled) {
for (int i = 0; i < overlap; i++)
out_samples[i] = out_dist_frame[i] / 1.5f;
} else {
memcpy(out_samples, in_frame, sizeof(*out_samples) * overlap);
}
}
static int drc_channel(AVFilterContext *ctx, AVFrame *in, AVFrame *out, int ch)
{
AudioDRCContext *s = ctx->priv;
const float *src = (const float *)in->extended_data[ch];
float *in_buffer = (float *)s->in_buffer->extended_data[ch];
float *dst = (float *)out->extended_data[ch];
memcpy(in_buffer, src, sizeof(*in_buffer) * s->overlap);
feed(ctx, ch, in_buffer, dst,
(float *)(s->in_frame->extended_data[ch]),
(float *)(s->out_dist_frame->extended_data[ch]),
(float *)(s->windowed_frame->extended_data[ch]),
(float *)(s->drc_frame->extended_data[ch]),
(float *)(s->spectrum_buf->extended_data[ch]),
(float *)(s->energy->extended_data[ch]),
(float *)(s->target_gain->extended_data[ch]),
(float *)(s->envelope->extended_data[ch]),
(float *)(s->factors->extended_data[ch]));
return 0;
}
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,
};

1
libavfilter/allfilters.c
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@ -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;

2
libavfilter/version.h
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@ -31,7 +31,7 @@
#include "version_major.h"
#define LIBAVFILTER_VERSION_MINOR 52
#define LIBAVFILTER_VERSION_MINOR 53
#define LIBAVFILTER_VERSION_MICRO 100

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