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FFmpeg/libavfilter/af_apsyclip.c
Jason Jang 306994b887 avfilter/af_apsyclip: increase protection for high frequencies 
This reduces sibilance distortion when sibilance and bass are
present at the same time. Bringing the protection of high
frequencies up to about the same level as for low frequencies
should also make the quality less dependent on the frequency
balance of the playback system.

Signed-off-by: Jason Jang <jcj83429@gmail.com>
2022-02-05 14:52:16 +01:00

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/*
* Copyright (c) 2014 - 2021 Jason Jang
* Copyright (c) 2021 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 "libavutil/opt.h"
#include "libavutil/tx.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
typedef struct AudioPsyClipContext {
const AVClass *class;
double level_in;
double level_out;
double clip_level;
double adaptive;
int auto_level;
int diff_only;
int iterations;
char *protections_str;
double *protections;
int num_psy_bins;
int fft_size;
int overlap;
int channels;
int spread_table_rows;
int *spread_table_index;
int (*spread_table_range)[2];
float *window, *inv_window, *spread_table, *margin_curve;
AVFrame *in;
AVFrame *in_buffer;
AVFrame *in_frame;
AVFrame *out_dist_frame;
AVFrame *windowed_frame;
AVFrame *clipping_delta;
AVFrame *spectrum_buf;
AVFrame *mask_curve;
AVTXContext **tx_ctx;
av_tx_fn tx_fn;
AVTXContext **itx_ctx;
av_tx_fn itx_fn;
} AudioPsyClipContext;
#define OFFSET(x) offsetof(AudioPsyClipContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption apsyclip_options[] = {
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
{ "clip", "set clip level", OFFSET(clip_level), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 1, FLAGS },
{ "diff", "enable difference", OFFSET(diff_only), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{ "adaptive", "set adaptive distortion", OFFSET(adaptive), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
{ "iterations", "set iterations", OFFSET(iterations), AV_OPT_TYPE_INT, {.i64=10}, 1, 20, FLAGS },
{ "level", "set auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{NULL}
};
AVFILTER_DEFINE_CLASS(apsyclip);
static void generate_hann_window(float *window, float *inv_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;
// 1/window to calculate unwindowed peak.
inv_window[i] = value > 0.1f ? 1.f / value : 0.f;
}
}
static void set_margin_curve(AudioPsyClipContext *s,
const int (*points)[2], int num_points, int sample_rate)
{
int j = 0;
s->margin_curve[0] = points[0][1];
for (int i = 0; i < num_points - 1; i++) {
while (j < s->fft_size / 2 + 1 && j * sample_rate / s->fft_size < points[i + 1][0]) {
// linearly interpolate between points
int binHz = j * sample_rate / s->fft_size;
s->margin_curve[j] = points[i][1] + (binHz - points[i][0]) * (points[i + 1][1] - points[i][1]) / (points[i + 1][0] - points[i][0]);
j++;
}
}
// handle bins after the last point
while (j < s->fft_size / 2 + 1) {
s->margin_curve[j] = points[num_points - 1][1];
j++;
}
// convert margin curve to linear amplitude scale
for (j = 0; j < s->fft_size / 2 + 1; j++)
s->margin_curve[j] = powf(10.f, s->margin_curve[j] / 20.f);
}
static void generate_spread_table(AudioPsyClipContext *s)
{
// Calculate tent-shape function in log-log scale.
// As an optimization, only consider bins close to "bin"
// This reduces the number of multiplications needed in calculate_mask_curve
// The masking contribution at faraway bins is negligeable
// Another optimization to save memory and speed up the calculation of the
// spread table is to calculate and store only 2 spread functions per
// octave, and reuse the same spread function for multiple bins.
int table_index = 0;
int bin = 0;
int increment = 1;
while (bin < s->num_psy_bins) {
float sum = 0;
int base_idx = table_index * s->num_psy_bins;
int start_bin = bin * 3 / 4;
int end_bin = FFMIN(s->num_psy_bins, ((bin + 1) * 4 + 2) / 3);
int next_bin;
for (int j = start_bin; j < end_bin; j++) {
// add 0.5 so i=0 doesn't get log(0)
float rel_idx_log = FFABS(logf((j + 0.5f) / (bin + 0.5f)));
float value;
if (j >= bin) {
// mask up
value = expf(-rel_idx_log * 40.f);
} else {
// mask down
value = expf(-rel_idx_log * 80.f);
}
// the spreading function is centred in the row
sum += value;
s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] = value;
}
// now normalize it
for (int j = start_bin; j < end_bin; j++) {
s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] /= sum;
}
s->spread_table_range[table_index][0] = start_bin - bin;
s->spread_table_range[table_index][1] = end_bin - bin;
if (bin <= 1) {
next_bin = bin + 1;
} else {
if ((bin & (bin - 1)) == 0) {
// power of 2
increment = bin / 2;
}
next_bin = bin + increment;
}
// set bins between "bin" and "next_bin" to use this table_index
for (int i = bin; i < next_bin; i++)
s->spread_table_index[i] = table_index;
bin = next_bin;
table_index++;
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioPsyClipContext *s = ctx->priv;
static const int points[][2] = { {0,14}, {125,14}, {250,16}, {500,18}, {1000,20}, {2000,20}, {4000,20}, {8000,17}, {16000,14}, {20000,-10} };
static const int num_points = 10;
float scale;
int ret;
s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
s->overlap = s->fft_size / 4;
// The psy masking calculation is O(n^2),
// so skip it for frequencies not covered by base sampling rantes (i.e. 44k)
if (inlink->sample_rate <= 50000) {
s->num_psy_bins = s->fft_size / 2;
} else if (inlink->sample_rate <= 100000) {
s->num_psy_bins = s->fft_size / 4;
} else {
s->num_psy_bins = s->fft_size / 8;
}
s->window = av_calloc(s->fft_size, sizeof(*s->window));
s->inv_window = av_calloc(s->fft_size, sizeof(*s->inv_window));
if (!s->window || !s->inv_window)
return AVERROR(ENOMEM);
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->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->clipping_delta = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->mask_curve = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
if (!s->in_buffer || !s->in_frame ||
!s->out_dist_frame || !s->windowed_frame ||
!s->clipping_delta || !s->spectrum_buf || !s->mask_curve)
return AVERROR(ENOMEM);
generate_hann_window(s->window, s->inv_window, s->fft_size);
s->margin_curve = av_calloc(s->fft_size / 2 + 1, sizeof(*s->margin_curve));
if (!s->margin_curve)
return AVERROR(ENOMEM);
s->spread_table_rows = av_log2(s->num_psy_bins) * 2;
s->spread_table = av_calloc(s->spread_table_rows * s->num_psy_bins, sizeof(*s->spread_table));
if (!s->spread_table)
return AVERROR(ENOMEM);
s->spread_table_range = av_calloc(s->spread_table_rows * 2, sizeof(*s->spread_table_range));
if (!s->spread_table_range)
return AVERROR(ENOMEM);
s->spread_table_index = av_calloc(s->num_psy_bins, sizeof(*s->spread_table_index));
if (!s->spread_table_index)
return AVERROR(ENOMEM);
set_margin_curve(s, points, num_points, inlink->sample_rate);
generate_spread_table(s);
s->channels = inlink->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++) {
ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->fft_size, &scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_FFT, 1, s->fft_size, &scale, 0);
if (ret < 0)
return ret;
}
return 0;
}
static void apply_window(AudioPsyClipContext *s,
const float *in_frame, float *out_frame, const int add_to_out_frame)
{
const float *window = s->window;
for (int i = 0; i < s->fft_size; i++) {
if (add_to_out_frame) {
out_frame[i] += in_frame[i] * window[i];
} else {
out_frame[i] = in_frame[i] * window[i];
}
}
}
static void calculate_mask_curve(AudioPsyClipContext *s,
const float *spectrum, float *mask_curve)
{
for (int i = 0; i < s->fft_size / 2 + 1; i++)
mask_curve[i] = 0;
for (int i = 0; i < s->num_psy_bins; i++) {
int base_idx, start_bin, end_bin, table_idx;
float magnitude;
int range[2];
if (i == 0) {
magnitude = FFABS(spectrum[0]);
} else if (i == s->fft_size / 2) {
magnitude = FFABS(spectrum[s->fft_size]);
} else {
// Because the input signal is real, the + and - frequencies are redundant.
// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
}
table_idx = s->spread_table_index[i];
range[0] = s->spread_table_range[table_idx][0];
range[1] = s->spread_table_range[table_idx][1];
base_idx = table_idx * s->num_psy_bins;
start_bin = FFMAX(0, i + range[0]);
end_bin = FFMIN(s->num_psy_bins, i + range[1]);
for (int j = start_bin; j < end_bin; j++)
mask_curve[j] += s->spread_table[base_idx + s->num_psy_bins / 2 + j - i] * magnitude;
}
// for ultrasonic frequencies, skip the O(n^2) spread calculation and just copy the magnitude
for (int i = s->num_psy_bins; i < s->fft_size / 2 + 1; i++) {
float magnitude;
if (i == s->fft_size / 2) {
magnitude = FFABS(spectrum[s->fft_size]);
} else {
// Because the input signal is real, the + and - frequencies are redundant.
// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
}
mask_curve[i] = magnitude;
}
for (int i = 0; i < s->fft_size / 2 + 1; i++)
mask_curve[i] = mask_curve[i] / s->margin_curve[i];
}
static void clip_to_window(AudioPsyClipContext *s,
const float *windowed_frame, float *clipping_delta, float delta_boost)
{
const float *window = s->window;
for (int i = 0; i < s->fft_size; i++) {
const float limit = s->clip_level * window[i];
const float effective_value = windowed_frame[i] + clipping_delta[i];
if (effective_value > limit) {
clipping_delta[i] += (limit - effective_value) * delta_boost;
} else if (effective_value < -limit) {
clipping_delta[i] += (-limit - effective_value) * delta_boost;
}
}
}
static void limit_clip_spectrum(AudioPsyClipContext *s,
float *clip_spectrum, const float *mask_curve)
{
// bin 0
float relative_distortion_level = FFABS(clip_spectrum[0]) / mask_curve[0];
if (relative_distortion_level > 1.f)
clip_spectrum[0] /= relative_distortion_level;
// bin 1..N/2-1
for (int i = 1; i < s->fft_size / 2; i++) {
float real = clip_spectrum[i * 2];
float imag = clip_spectrum[i * 2 + 1];
// Because the input signal is real, the + and - frequencies are redundant.
// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
relative_distortion_level = hypotf(real, imag) * 2 / mask_curve[i];
if (relative_distortion_level > 1.0) {
clip_spectrum[i * 2] /= relative_distortion_level;
clip_spectrum[i * 2 + 1] /= relative_distortion_level;
clip_spectrum[s->fft_size * 2 - i * 2] /= relative_distortion_level;
clip_spectrum[s->fft_size * 2 - i * 2 + 1] /= relative_distortion_level;
}
}
// bin N/2
relative_distortion_level = FFABS(clip_spectrum[s->fft_size]) / mask_curve[s->fft_size / 2];
if (relative_distortion_level > 1.f)
clip_spectrum[s->fft_size] /= relative_distortion_level;
}
static void r2c(float *buffer, int size)
{
for (int i = size - 1; i >= 0; i--)
buffer[2 * i] = buffer[i];
for (int i = size - 1; i >= 0; i--)
buffer[2 * i + 1] = 0.f;
}
static void c2r(float *buffer, int size)
{
for (int i = 0; i < size; i++)
buffer[i] = buffer[2 * i];
for (int i = 0; i < size; i++)
buffer[i + size] = 0.f;
}
static void feed(AVFilterContext *ctx, int ch,
const float *in_samples, float *out_samples, int diff_only,
float *in_frame, float *out_dist_frame,
float *windowed_frame, float *clipping_delta,
float *spectrum_buf, float *mask_curve)
{
AudioPsyClipContext *s = ctx->priv;
const float clip_level_inv = 1.f / s->clip_level;
const float level_out = s->level_out;
float orig_peak = 0;
float peak;
// shift in/out buffers
for (int i = 0; i < s->fft_size - s->overlap; i++) {
in_frame[i] = in_frame[i + s->overlap];
out_dist_frame[i] = out_dist_frame[i + s->overlap];
}
for (int i = 0; i < s->overlap; i++) {
in_frame[i + s->fft_size - s->overlap] = in_samples[i];
out_dist_frame[i + s->fft_size - s->overlap] = 0.f;
}
apply_window(s, in_frame, windowed_frame, 0);
r2c(windowed_frame, s->fft_size);
s->tx_fn(s->tx_ctx[ch], spectrum_buf, windowed_frame, sizeof(float));
c2r(windowed_frame, s->fft_size);
calculate_mask_curve(s, spectrum_buf, mask_curve);
// It would be easier to calculate the peak from the unwindowed input.
// This is just for consistency with the clipped peak calculateion
// because the inv_window zeros out samples on the edge of the window.
for (int i = 0; i < s->fft_size; i++)
orig_peak = FFMAX(orig_peak, FFABS(windowed_frame[i] * s->inv_window[i]));
orig_peak *= clip_level_inv;
peak = orig_peak;
// clear clipping_delta
for (int i = 0; i < s->fft_size * 2; i++)
clipping_delta[i] = 0.f;
// repeat clipping-filtering process a few times to control both the peaks and the spectrum
for (int i = 0; i < s->iterations; i++) {
float mask_curve_shift = 1.122f; // 1.122 is 1dB
// The last 1/3 of rounds have boosted delta to help reach the peak target faster
float delta_boost = 1.f;
if (i >= s->iterations - s->iterations / 3) {
// boosting the delta when largs peaks are still present is dangerous
if (peak < 2.f)
delta_boost = 2.f;
}
clip_to_window(s, windowed_frame, clipping_delta, delta_boost);
r2c(clipping_delta, s->fft_size);
s->tx_fn(s->tx_ctx[ch], spectrum_buf, clipping_delta, sizeof(float));
limit_clip_spectrum(s, spectrum_buf, mask_curve);
s->itx_fn(s->itx_ctx[ch], clipping_delta, spectrum_buf, sizeof(float));
c2r(clipping_delta, s->fft_size);
for (int i = 0; i < s->fft_size; i++)
clipping_delta[i] /= s->fft_size;
peak = 0;
for (int i = 0; i < s->fft_size; i++)
peak = FFMAX(peak, FFABS((windowed_frame[i] + clipping_delta[i]) * s->inv_window[i]));
peak *= clip_level_inv;
// Automatically adjust mask_curve as necessary to reach peak target
if (orig_peak > 1.f && peak > 1.f) {
float diff_achieved = orig_peak - peak;
if (i + 1 < s->iterations - s->iterations / 3 && diff_achieved > 0) {
float diff_needed = orig_peak - 1.f;
float diff_ratio = diff_needed / diff_achieved;
// If a good amount of peak reduction was already achieved,
// don't shift the mask_curve by the full peak value
// On the other hand, if only a little peak reduction was achieved,
// don't shift the mask_curve by the enormous diff_ratio.
diff_ratio = FFMIN(diff_ratio, peak);
mask_curve_shift = FFMAX(mask_curve_shift, diff_ratio);
} else {
// If the peak got higher than the input or we are in the last 1/3 rounds,
// go back to the heavy-handed peak heuristic.
mask_curve_shift = FFMAX(mask_curve_shift, peak);
}
}
mask_curve_shift = 1.f + (mask_curve_shift - 1.f) * s->adaptive;
// Be less strict in the next iteration.
// This helps with peak control.
for (int i = 0; i < s->fft_size / 2 + 1; i++)
mask_curve[i] *= mask_curve_shift;
}
// do overlap & add
apply_window(s, clipping_delta, out_dist_frame, 1);
for (int i = 0; i < s->overlap; i++) {
// 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
if (!ctx->is_disabled) {
out_samples[i] = out_dist_frame[i] / 1.5f;
if (!diff_only)
out_samples[i] += in_frame[i];
if (s->auto_level)
out_samples[i] *= clip_level_inv;
out_samples[i] *= level_out;
} else {
out_samples[i] = in_frame[i];
}
}
}
static int psy_channel(AVFilterContext *ctx, AVFrame *in, AVFrame *out, int ch)
{
AudioPsyClipContext *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];
for (int n = 0; n < s->overlap; n++)
in_buffer[n] = src[n] * s->level_in;
feed(ctx, ch, in_buffer, dst, s->diff_only,
(float *)(s->in_frame->extended_data[ch]),
(float *)(s->out_dist_frame->extended_data[ch]),
(float *)(s->windowed_frame->extended_data[ch]),
(float *)(s->clipping_delta->extended_data[ch]),
(float *)(s->spectrum_buf->extended_data[ch]),
(float *)(s->mask_curve->extended_data[ch]));
return 0;
}
static int psy_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioPsyClipContext *s = ctx->priv;
AVFrame *out = arg;
const int start = (out->channels * jobnr) / nb_jobs;
const int end = (out->channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++)
psy_channel(ctx, s->in, out, ch);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioPsyClipContext *s = ctx->priv;
AVFrame *out;
int ret;
out = ff_get_audio_buffer(outlink, s->overlap);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
s->in = in;
ff_filter_execute(ctx, psy_channels, out, NULL,
FFMIN(outlink->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];
AudioPsyClipContext *s = ctx->priv;
AVFrame *in = NULL;
int ret = 0, status;
int64_t pts;
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) {
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)
{
AudioPsyClipContext *s = ctx->priv;
av_freep(&s->window);
av_freep(&s->inv_window);
av_freep(&s->spread_table);
av_freep(&s->spread_table_range);
av_freep(&s->spread_table_index);
av_freep(&s->margin_curve);
av_frame_free(&s->in_buffer);
av_frame_free(&s->in_frame);
av_frame_free(&s->out_dist_frame);
av_frame_free(&s->windowed_frame);
av_frame_free(&s->clipping_delta);
av_frame_free(&s->spectrum_buf);
av_frame_free(&s->mask_curve);
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 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_apsyclip = {
.name = "apsyclip",
.description = NULL_IF_CONFIG_SMALL("Audio Psychoacoustic Clipper."),
.priv_size = sizeof(AudioPsyClipContext),
.priv_class = &apsyclip_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 = ff_filter_process_command,
};
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