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FFmpeg/libavfilter/af_adrc.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

503 lines
16 KiB
C

/*
* 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/mem.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,
},
};
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(ff_audio_default_filterpad),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.activate = activate,
.process_command = process_command,
};