@@ -2281,6 +2281,21 @@ Default value is 1.0.
This filter supports the all above options as @ref{commands}.
+@section apsyclip
+Apply Psychoacoustic clipper to audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item level_in
+@item level_out
+@item clip_level
+@item diff_only
+@item adaptive
+@item iterations
+@item auto_level
+@end table
+
@section apulsator
Audio pulsator is something between an autopanner and a tremolo.
@@ -72,6 +72,7 @@ OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
+OBJS-$(CONFIG_APSYCLIP_FILTER) += af_apsyclip.o
OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
new file mode 100644
@@ -0,0 +1,645 @@
+/*
+ * 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_distortion;
+ int auto_level;
+ int diff_only;
+ int iterations;
+ char *protections_str;
+ double *protections;
+
+ int num_psy_bins;
+ int fft_size;
+ int overlap;
+
+ int spread_table_rows;
+ int *spread_table_index;
+ int (*spread_table_range)[2];
+ float *window, *inv_window, *spread_table, *margin_curve;
+
+ AVFrame *in_buffer;
+ AVFrame *out_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 A AV_OPT_FLAG_AUDIO_PARAM
+#define F AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption apsyclip_options[] = {
+ { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
+ { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
+ { "clip_level", "set clip level", OFFSET(clip_level), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 1, A|F },
+ { "diff_only", "set hear difference", OFFSET(diff_only), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A|F },
+ { "adaptive", "set adaptive distortion", OFFSET(adaptive_distortion), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A|F },
+ { "iterations", "set iterations", OFFSET(iterations), AV_OPT_TYPE_INT, {.i64=10}, 0, 20, A|F },
+ { "auto_level", "set auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL,{.i64=1}, 0, 1, A|F },
+ {NULL}
+};
+
+AVFILTER_DEFINE_CLASS(apsyclip);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP,
+ 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 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 > 0.01f ? 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}, {8000,15}, {16000,5}, {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->out_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->out_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);
+
+ ret = av_tx_init(&s->tx_ctx, &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, &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++) {
+ float magnitude;
+ int table_idx;
+ int range[2];
+
+ if (i == 0) {
+ magnitude = FFABS(spectrum[0]);
+ } else if (i == s->fft_size / 2) {
+ magnitude = FFABS(spectrum[1]);
+ } else {
+ // although the negative frequencies are omitted because they are redundant,
+ // the magnitude of the positive frequencies are not doubled.
+ // 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];
+ int base_idx = table_idx * s->num_psy_bins;
+ int start_bin = FFMAX(0, i + range[0]);
+ int 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[1]);
+ } else {
+ // although the negative frequencies are omitted because they are redundant,
+ // the magnitude of the positive frequencies are not doubled.
+ // 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++) {
+ float limit = s->clip_level * window[i];
+ 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];
+ // although the negative frequencies are omitted because they are redundant,
+ // the magnitude of the positive frequencies are not doubled.
+ // 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;
+ }
+ }
+ // bin N/2
+ relative_distortion_level = FFABS(clip_spectrum[1]) / mask_curve[s->fft_size / 2];
+ if (relative_distortion_level > 1.f)
+ clip_spectrum[1] /= 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(AudioPsyClipContext *s,
+ 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)
+{
+ 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, 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 /= s->clip_level;
+ 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, spectrum_buf, clipping_delta, sizeof(float));
+
+ limit_clip_spectrum(s, spectrum_buf, mask_curve);
+
+ s->itx_fn(s->itx_ctx, 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 /= s->clip_level;
+
+ // 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_distortion;
+
+ // 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++) {
+ out_samples[i] = out_dist_frame[i] / 1.5f;
+ // 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
+ if (!diff_only)
+ out_samples[i] += in_frame[i];
+ }
+}
+
+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;
+ }
+
+ out->pts = in->pts;
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ 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(s, 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]));
+ }
+
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ goto fail;
+
+fail:
+ av_frame_free(&in);
+ 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_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->out_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);
+
+ av_tx_uninit(&s->tx_ctx);
+ av_tx_uninit(&s->itx_ctx);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+const AVFilter ff_af_apsyclip = {
+ .name = "apsyclip",
+ .description = NULL_IF_CONFIG_SMALL("Audio Psychoacoustic Clipper."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioPsyClipContext),
+ .priv_class = &apsyclip_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .activate = activate,
+};
@@ -65,6 +65,7 @@ extern const AVFilter ff_af_apad;
extern const AVFilter ff_af_aperms;
extern const AVFilter ff_af_aphaser;
extern const AVFilter ff_af_aphaseshift;
+extern const AVFilter ff_af_apsyclip;
extern const AVFilter ff_af_apulsator;
extern const AVFilter ff_af_arealtime;
extern const AVFilter ff_af_aresample;
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- doc/filters.texi | 15 + libavfilter/Makefile | 1 + libavfilter/af_apsyclip.c | 645 ++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 662 insertions(+) create mode 100644 libavfilter/af_apsyclip.c