diff --git a/Changelog b/Changelog index 06cb2b2190..d0b1a9724e 100644 --- a/Changelog +++ b/Changelog @@ -19,6 +19,7 @@ version : - swscale slice threading - MSN Siren decoder - scharr video filter +- apsyclip audio filter version 4.4: diff --git a/doc/filters.texi b/doc/filters.texi index aabfaccfc3..b3acc88ef2 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2363,6 +2363,42 @@ Default value is 8. This filter supports the all above options as @ref{commands}. +@section apsyclip +Apply Psychoacoustic clipper to input audio stream. + +The filter accepts the following options: + +@table @option +@item level_in +Set input gain. By default it is 1. Range is [0.015625 - 64]. + +@item level_out +Set output gain. By default it is 1. Range is [0.015625 - 64]. + +@item clip +Set the clipping start value. Default value is 0dBFS or 1. + +@item diff +Output only difference samples, useful to hear introduced distortions. +By default is disabled. + +@item adaptive +Set strenght of adaptive distortion applied. Default value is 0.5. +Allowed range is from 0 to 1. + +@item iterations +Set number of iterations of psychoacoustic clipper. +Allowed range is from 1 to 20. Default value is 10. + +@item level +Auto level output signal. Default is disabled. +This normalizes audio back to 0dBFS if enabled. +@end table + +@subsection Commands + +This filter supports the all above options as @ref{commands}. + @section apulsator Audio pulsator is something between an autopanner and a tremolo. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 9d71aa6b3c..f059f3fef8 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -74,6 +74,7 @@ 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 diff --git a/libavfilter/af_apsyclip.c b/libavfilter/af_apsyclip.c new file mode 100644 index 0000000000..6fec4f52c2 --- /dev/null +++ b/libavfilter/af_apsyclip.c @@ -0,0 +1,679 @@ +/* + * 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 int query_formats(AVFilterContext *ctx) +{ + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + ret = ff_set_common_all_channel_counts(ctx); + if (ret < 0) + return ret; + + ret = ff_set_common_formats_from_list(ctx, sample_fmts); + if (ret < 0) + return ret; + + return ff_set_common_all_samplerates(ctx); +} + +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->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++) { + 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++) { + 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]; + // 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(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."), + .query_formats = query_formats, + .priv_size = sizeof(AudioPsyClipContext), + .priv_class = &apsyclip_class, + .uninit = uninit, + FILTER_INPUTS(inputs), + FILTER_OUTPUTS(outputs), + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | + AVFILTER_FLAG_SLICE_THREADS, + .activate = activate, + .process_command = ff_filter_process_command, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 9313a0674b..ddd6404228 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -67,6 +67,7 @@ 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; diff --git a/libavfilter/version.h b/libavfilter/version.h index 306bb62ff4..24b59acde6 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 8 -#define LIBAVFILTER_VERSION_MINOR 8 +#define LIBAVFILTER_VERSION_MINOR 9 #define LIBAVFILTER_VERSION_MICRO 100