1435 lines
44 KiB
C
1435 lines
44 KiB
C
/*
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* Copyright (c) 2018 The FFmpeg Project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <float.h>
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "libavcodec/avfft.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#include "filters.h"
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#define C (M_LN10 * 0.1)
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#define RATIO 0.98
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#define RRATIO (1.0 - RATIO)
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enum OutModes {
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IN_MODE,
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OUT_MODE,
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NOISE_MODE,
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NB_MODES
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};
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enum NoiseType {
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WHITE_NOISE,
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VINYL_NOISE,
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SHELLAC_NOISE,
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CUSTOM_NOISE,
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NB_NOISE
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};
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typedef struct DeNoiseChannel {
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int band_noise[15];
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double noise_band_auto_var[15];
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double noise_band_sample[15];
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double *amt;
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double *band_amt;
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double *band_excit;
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double *gain;
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double *prior;
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double *prior_band_excit;
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double *clean_data;
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double *noisy_data;
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double *out_samples;
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double *spread_function;
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double *abs_var;
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double *rel_var;
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double *min_abs_var;
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FFTComplex *fft_data;
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FFTContext *fft, *ifft;
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double noise_band_norm[15];
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double noise_band_avr[15];
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double noise_band_avi[15];
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double noise_band_var[15];
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double sfm_threshold;
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double sfm_alpha;
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double sfm_results[3];
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int sfm_fail_flags[512];
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int sfm_fail_total;
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} DeNoiseChannel;
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typedef struct AudioFFTDeNoiseContext {
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const AVClass *class;
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float noise_reduction;
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float noise_floor;
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int noise_type;
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char *band_noise_str;
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float residual_floor;
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int track_noise;
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int track_residual;
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int output_mode;
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float last_residual_floor;
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float last_noise_floor;
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float last_noise_reduction;
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float last_noise_balance;
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int64_t block_count;
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int64_t pts;
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int channels;
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int sample_noise;
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int sample_noise_start;
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int sample_noise_end;
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float sample_rate;
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int buffer_length;
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int fft_length;
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int fft_length2;
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int bin_count;
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int window_length;
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int sample_advance;
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int number_of_bands;
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int band_centre[15];
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int *bin2band;
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double *window;
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double *band_alpha;
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double *band_beta;
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DeNoiseChannel *dnch;
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double max_gain;
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double max_var;
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double gain_scale;
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double window_weight;
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double floor;
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double sample_floor;
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double auto_floor;
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int noise_band_edge[17];
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int noise_band_count;
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double matrix_a[25];
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double vector_b[5];
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double matrix_b[75];
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double matrix_c[75];
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AVAudioFifo *fifo;
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} AudioFFTDeNoiseContext;
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#define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption afftdn_options[] = {
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{ "nr", "set the noise reduction", OFFSET(noise_reduction), AV_OPT_TYPE_FLOAT, {.dbl = 12}, .01, 97, AFR },
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{ "nf", "set the noise floor", OFFSET(noise_floor), AV_OPT_TYPE_FLOAT, {.dbl =-50}, -80,-20, AFR },
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{ "nt", "set the noise type", OFFSET(noise_type), AV_OPT_TYPE_INT, {.i64 = WHITE_NOISE}, WHITE_NOISE, NB_NOISE-1, AF, "type" },
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{ "w", "white noise", 0, AV_OPT_TYPE_CONST, {.i64 = WHITE_NOISE}, 0, 0, AF, "type" },
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{ "v", "vinyl noise", 0, AV_OPT_TYPE_CONST, {.i64 = VINYL_NOISE}, 0, 0, AF, "type" },
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{ "s", "shellac noise", 0, AV_OPT_TYPE_CONST, {.i64 = SHELLAC_NOISE}, 0, 0, AF, "type" },
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{ "c", "custom noise", 0, AV_OPT_TYPE_CONST, {.i64 = CUSTOM_NOISE}, 0, 0, AF, "type" },
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{ "bn", "set the custom bands noise", OFFSET(band_noise_str), AV_OPT_TYPE_STRING, {.str = 0}, 0, 0, AF },
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{ "rf", "set the residual floor", OFFSET(residual_floor), AV_OPT_TYPE_FLOAT, {.dbl =-38}, -80,-20, AFR },
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{ "tn", "track noise", OFFSET(track_noise), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
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{ "tr", "track residual", OFFSET(track_residual), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AFR },
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{ "om", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64 = OUT_MODE}, 0, NB_MODES-1, AFR, "mode" },
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{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64 = IN_MODE}, 0, 0, AFR, "mode" },
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{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64 = OUT_MODE}, 0, 0, AFR, "mode" },
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{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64 = NOISE_MODE}, 0, 0, AFR, "mode" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(afftdn);
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static int get_band_noise(AudioFFTDeNoiseContext *s,
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int band, double a,
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double b, double c)
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{
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double d1, d2, d3;
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d1 = a / s->band_centre[band];
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d1 = 10.0 * log(1.0 + d1 * d1) / M_LN10;
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d2 = b / s->band_centre[band];
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d2 = 10.0 * log(1.0 + d2 * d2) / M_LN10;
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d3 = s->band_centre[band] / c;
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d3 = 10.0 * log(1.0 + d3 * d3) / M_LN10;
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return lrint(-d1 + d2 - d3);
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}
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static void factor(double *array, int size)
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{
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for (int i = 0; i < size - 1; i++) {
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for (int j = i + 1; j < size; j++) {
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double d = array[j + i * size] / array[i + i * size];
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array[j + i * size] = d;
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for (int k = i + 1; k < size; k++) {
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array[j + k * size] -= d * array[i + k * size];
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}
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}
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}
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}
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static void solve(double *matrix, double *vector, int size)
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{
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for (int i = 0; i < size - 1; i++) {
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for (int j = i + 1; j < size; j++) {
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double d = matrix[j + i * size];
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vector[j] -= d * vector[i];
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}
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}
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vector[size - 1] /= matrix[size * size - 1];
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for (int i = size - 2; i >= 0; i--) {
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double d = vector[i];
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for (int j = i + 1; j < size; j++)
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d -= matrix[i + j * size] * vector[j];
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vector[i] = d / matrix[i + i * size];
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}
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}
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static int process_get_band_noise(AudioFFTDeNoiseContext *s,
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DeNoiseChannel *dnch,
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int band)
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{
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double product, sum, f;
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int i = 0;
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if (band < 15)
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return dnch->band_noise[band];
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for (int j = 0; j < 5; j++) {
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sum = 0.0;
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for (int k = 0; k < 15; k++)
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sum += s->matrix_b[i++] * dnch->band_noise[k];
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s->vector_b[j] = sum;
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}
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solve(s->matrix_a, s->vector_b, 5);
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f = (0.5 * s->sample_rate) / s->band_centre[14];
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f = 15.0 + log(f / 1.5) / log(1.5);
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sum = 0.0;
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product = 1.0;
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for (int j = 0; j < 5; j++) {
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sum += product * s->vector_b[j];
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product *= f;
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}
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return lrint(sum);
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}
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static void calculate_sfm(AudioFFTDeNoiseContext *s,
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DeNoiseChannel *dnch,
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int start, int end)
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{
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double d1 = 0.0, d2 = 1.0;
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int i = 0, j = 0;
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for (int k = start; k < end; k++) {
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if (dnch->noisy_data[k] > s->sample_floor) {
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j++;
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d1 += dnch->noisy_data[k];
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d2 *= dnch->noisy_data[k];
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if (d2 > 1.0E100) {
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d2 *= 1.0E-100;
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i++;
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} else if (d2 < 1.0E-100) {
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d2 *= 1.0E100;
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i--;
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}
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}
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}
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if (j > 1) {
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d1 /= j;
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dnch->sfm_results[0] = d1;
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d2 = log(d2) + 230.2585 * i;
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d2 /= j;
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d1 = log(d1);
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dnch->sfm_results[1] = d1;
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dnch->sfm_results[2] = d1 - d2;
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} else {
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dnch->sfm_results[0] = s->auto_floor;
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dnch->sfm_results[1] = dnch->sfm_threshold;
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dnch->sfm_results[2] = dnch->sfm_threshold;
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}
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}
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static double limit_gain(double a, double b)
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{
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if (a > 1.0)
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return (b * a - 1.0) / (b + a - 2.0);
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if (a < 1.0)
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return (b * a - 2.0 * a + 1.0) / (b - a);
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return 1.0;
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}
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static void process_frame(AudioFFTDeNoiseContext *s, DeNoiseChannel *dnch,
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FFTComplex *fft_data,
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double *prior, double *prior_band_excit, int track_noise)
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{
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double d1, d2, d3, gain;
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int n, i1;
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d1 = fft_data[0].re * fft_data[0].re;
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dnch->noisy_data[0] = d1;
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d2 = d1 / dnch->abs_var[0];
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d3 = RATIO * prior[0] + RRATIO * fmax(d2 - 1.0, 0.0);
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gain = d3 / (1.0 + d3);
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gain *= (gain + M_PI_4 / fmax(d2, 1.0E-6));
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prior[0] = (d2 * gain);
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dnch->clean_data[0] = (d1 * gain);
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gain = sqrt(gain);
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dnch->gain[0] = gain;
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n = 0;
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for (int i = 1; i < s->fft_length2; i++) {
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d1 = fft_data[i].re * fft_data[i].re + fft_data[i].im * fft_data[i].im;
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if (d1 > s->sample_floor)
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n = i;
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dnch->noisy_data[i] = d1;
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d2 = d1 / dnch->abs_var[i];
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d3 = RATIO * prior[i] + RRATIO * fmax(d2 - 1.0, 0.0);
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gain = d3 / (1.0 + d3);
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gain *= (gain + M_PI_4 / fmax(d2, 1.0E-6));
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prior[i] = d2 * gain;
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dnch->clean_data[i] = d1 * gain;
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gain = sqrt(gain);
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dnch->gain[i] = gain;
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}
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d1 = fft_data[0].im * fft_data[0].im;
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if (d1 > s->sample_floor)
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n = s->fft_length2;
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dnch->noisy_data[s->fft_length2] = d1;
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d2 = d1 / dnch->abs_var[s->fft_length2];
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d3 = RATIO * prior[s->fft_length2] + RRATIO * fmax(d2 - 1.0, 0.0);
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gain = d3 / (1.0 + d3);
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gain *= gain + M_PI_4 / fmax(d2, 1.0E-6);
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prior[s->fft_length2] = d2 * gain;
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dnch->clean_data[s->fft_length2] = d1 * gain;
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gain = sqrt(gain);
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dnch->gain[s->fft_length2] = gain;
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if (n > s->fft_length2 - 2) {
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n = s->bin_count;
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i1 = s->noise_band_count;
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} else {
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i1 = 0;
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for (int i = 0; i <= s->noise_band_count; i++) {
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if (n > 1.1 * s->noise_band_edge[i]) {
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i1 = i;
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}
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}
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}
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if (track_noise && (i1 > s->noise_band_count / 2)) {
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int j = FFMIN(n, s->noise_band_edge[i1]);
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int m = 3, k;
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for (k = i1 - 1; k >= 0; k--) {
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int i = s->noise_band_edge[k];
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calculate_sfm(s, dnch, i, j);
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dnch->noise_band_sample[k] = dnch->sfm_results[0];
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if (dnch->sfm_results[2] + 0.013 * m * fmax(0.0, dnch->sfm_results[1] - 20.53) >= dnch->sfm_threshold) {
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break;
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}
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j = i;
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m++;
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}
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if (k < i1 - 1) {
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double sum = 0.0, min, max;
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int i;
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for (i = i1 - 1; i > k; i--) {
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min = log(dnch->noise_band_sample[i] / dnch->noise_band_auto_var[i]);
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sum += min;
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}
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i = i1 - k - 1;
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if (i < 5) {
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min = 3.0E-4 * i * i;
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} else {
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min = 3.0E-4 * (8 * i - 16);
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}
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if (i < 3) {
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max = 2.0E-4 * i * i;
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} else {
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max = 2.0E-4 * (4 * i - 4);
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}
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if (s->track_residual) {
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if (s->last_noise_floor > s->last_residual_floor + 9) {
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min *= 0.5;
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max *= 0.75;
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} else if (s->last_noise_floor > s->last_residual_floor + 6) {
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min *= 0.4;
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max *= 1.0;
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} else if (s->last_noise_floor > s->last_residual_floor + 4) {
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min *= 0.3;
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max *= 1.3;
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} else if (s->last_noise_floor > s->last_residual_floor + 2) {
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min *= 0.2;
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max *= 1.6;
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} else if (s->last_noise_floor > s->last_residual_floor) {
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min *= 0.1;
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max *= 2.0;
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} else {
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min = 0.0;
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max *= 2.5;
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}
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}
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sum = av_clipd(sum, -min, max);
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sum = exp(sum);
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for (int i = 0; i < 15; i++)
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dnch->noise_band_auto_var[i] *= sum;
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} else if (dnch->sfm_results[2] >= dnch->sfm_threshold) {
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dnch->sfm_fail_flags[s->block_count & 0x1FF] = 1;
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dnch->sfm_fail_total += 1;
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}
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}
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for (int i = 0; i < s->number_of_bands; i++) {
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dnch->band_excit[i] = 0.0;
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dnch->band_amt[i] = 0.0;
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}
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for (int i = 0; i < s->bin_count; i++) {
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dnch->band_excit[s->bin2band[i]] += dnch->clean_data[i];
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}
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for (int i = 0; i < s->number_of_bands; i++) {
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dnch->band_excit[i] = fmax(dnch->band_excit[i],
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s->band_alpha[i] * dnch->band_excit[i] +
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s->band_beta[i] * prior_band_excit[i]);
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prior_band_excit[i] = dnch->band_excit[i];
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}
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for (int j = 0, i = 0; j < s->number_of_bands; j++) {
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for (int k = 0; k < s->number_of_bands; k++) {
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dnch->band_amt[j] += dnch->spread_function[i++] * dnch->band_excit[k];
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}
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}
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for (int i = 0; i < s->bin_count; i++)
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dnch->amt[i] = dnch->band_amt[s->bin2band[i]];
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if (dnch->amt[0] > dnch->abs_var[0]) {
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dnch->gain[0] = 1.0;
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} else if (dnch->amt[0] > dnch->min_abs_var[0]) {
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double limit = sqrt(dnch->abs_var[0] / dnch->amt[0]);
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dnch->gain[0] = limit_gain(dnch->gain[0], limit);
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} else {
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dnch->gain[0] = limit_gain(dnch->gain[0], s->max_gain);
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}
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if (dnch->amt[s->fft_length2] > dnch->abs_var[s->fft_length2]) {
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dnch->gain[s->fft_length2] = 1.0;
|
|
} else if (dnch->amt[s->fft_length2] > dnch->min_abs_var[s->fft_length2]) {
|
|
double limit = sqrt(dnch->abs_var[s->fft_length2] / dnch->amt[s->fft_length2]);
|
|
dnch->gain[s->fft_length2] = limit_gain(dnch->gain[s->fft_length2], limit);
|
|
} else {
|
|
dnch->gain[s->fft_length2] = limit_gain(dnch->gain[s->fft_length2], s->max_gain);
|
|
}
|
|
|
|
for (int i = 1; i < s->fft_length2; i++) {
|
|
if (dnch->amt[i] > dnch->abs_var[i]) {
|
|
dnch->gain[i] = 1.0;
|
|
} else if (dnch->amt[i] > dnch->min_abs_var[i]) {
|
|
double limit = sqrt(dnch->abs_var[i] / dnch->amt[i]);
|
|
dnch->gain[i] = limit_gain(dnch->gain[i], limit);
|
|
} else {
|
|
dnch->gain[i] = limit_gain(dnch->gain[i], s->max_gain);
|
|
}
|
|
}
|
|
|
|
gain = dnch->gain[0];
|
|
dnch->clean_data[0] = (gain * gain * dnch->noisy_data[0]);
|
|
fft_data[0].re *= gain;
|
|
gain = dnch->gain[s->fft_length2];
|
|
dnch->clean_data[s->fft_length2] = (gain * gain * dnch->noisy_data[s->fft_length2]);
|
|
fft_data[0].im *= gain;
|
|
for (int i = 1; i < s->fft_length2; i++) {
|
|
gain = dnch->gain[i];
|
|
dnch->clean_data[i] = (gain * gain * dnch->noisy_data[i]);
|
|
fft_data[i].re *= gain;
|
|
fft_data[i].im *= gain;
|
|
}
|
|
}
|
|
|
|
static double freq2bark(double x)
|
|
{
|
|
double d = x / 7500.0;
|
|
|
|
return 13.0 * atan(7.6E-4 * x) + 3.5 * atan(d * d);
|
|
}
|
|
|
|
static int get_band_centre(AudioFFTDeNoiseContext *s, int band)
|
|
{
|
|
if (band == -1)
|
|
return lrint(s->band_centre[0] / 1.5);
|
|
|
|
return s->band_centre[band];
|
|
}
|
|
|
|
static int get_band_edge(AudioFFTDeNoiseContext *s, int band)
|
|
{
|
|
int i;
|
|
|
|
if (band == 15) {
|
|
i = lrint(s->band_centre[14] * 1.224745);
|
|
} else {
|
|
i = lrint(s->band_centre[band] / 1.224745);
|
|
}
|
|
|
|
return FFMIN(i, s->sample_rate / 2);
|
|
}
|
|
|
|
static void set_band_parameters(AudioFFTDeNoiseContext *s,
|
|
DeNoiseChannel *dnch)
|
|
{
|
|
double band_noise, d2, d3, d4, d5;
|
|
int i = 0, j = 0, k = 0;
|
|
|
|
d5 = 0.0;
|
|
band_noise = process_get_band_noise(s, dnch, 0);
|
|
for (int m = j; m <= s->fft_length2; m++) {
|
|
if (m == j) {
|
|
i = j;
|
|
d5 = band_noise;
|
|
if (k == 15) {
|
|
j = s->bin_count;
|
|
} else {
|
|
j = s->fft_length * get_band_centre(s, k) / s->sample_rate;
|
|
}
|
|
d2 = j - i;
|
|
band_noise = process_get_band_noise(s, dnch, k);
|
|
k++;
|
|
}
|
|
d3 = (j - m) / d2;
|
|
d4 = (m - i) / d2;
|
|
dnch->rel_var[m] = exp((d5 * d3 + band_noise * d4) * C);
|
|
}
|
|
dnch->rel_var[s->fft_length2] = exp(band_noise * C);
|
|
|
|
for (i = 0; i < 15; i++)
|
|
dnch->noise_band_auto_var[i] = s->max_var * exp((process_get_band_noise(s, dnch, i) - 2.0) * C);
|
|
|
|
for (i = 0; i <= s->fft_length2; i++) {
|
|
dnch->abs_var[i] = fmax(s->max_var * dnch->rel_var[i], 1.0);
|
|
dnch->min_abs_var[i] = s->gain_scale * dnch->abs_var[i];
|
|
}
|
|
}
|
|
|
|
static void read_custom_noise(AudioFFTDeNoiseContext *s, int ch)
|
|
{
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
char *p, *arg, *saveptr = NULL;
|
|
int i, ret, band_noise[15] = { 0 };
|
|
|
|
if (!s->band_noise_str)
|
|
return;
|
|
|
|
p = av_strdup(s->band_noise_str);
|
|
if (!p)
|
|
return;
|
|
|
|
for (i = 0; i < 15; i++) {
|
|
if (!(arg = av_strtok(p, "| ", &saveptr)))
|
|
break;
|
|
|
|
p = NULL;
|
|
|
|
ret = av_sscanf(arg, "%d", &band_noise[i]);
|
|
if (ret != 1) {
|
|
av_log(s, AV_LOG_ERROR, "Custom band noise must be integer.\n");
|
|
break;
|
|
}
|
|
|
|
band_noise[i] = av_clip(band_noise[i], -24, 24);
|
|
}
|
|
|
|
av_free(p);
|
|
memcpy(dnch->band_noise, band_noise, sizeof(band_noise));
|
|
}
|
|
|
|
static void set_parameters(AudioFFTDeNoiseContext *s)
|
|
{
|
|
if (s->last_noise_floor != s->noise_floor)
|
|
s->last_noise_floor = s->noise_floor;
|
|
|
|
if (s->track_residual)
|
|
s->last_noise_floor = fmaxf(s->last_noise_floor, s->residual_floor);
|
|
|
|
s->max_var = s->floor * exp((100.0 + s->last_noise_floor) * C);
|
|
|
|
if (s->track_residual) {
|
|
s->last_residual_floor = s->residual_floor;
|
|
s->last_noise_reduction = fmax(s->last_noise_floor - s->last_residual_floor, 0);
|
|
s->max_gain = exp(s->last_noise_reduction * (0.5 * C));
|
|
} else if (s->noise_reduction != s->last_noise_reduction) {
|
|
s->last_noise_reduction = s->noise_reduction;
|
|
s->last_residual_floor = av_clipf(s->last_noise_floor - s->last_noise_reduction, -80, -20);
|
|
s->max_gain = exp(s->last_noise_reduction * (0.5 * C));
|
|
}
|
|
|
|
s->gain_scale = 1.0 / (s->max_gain * s->max_gain);
|
|
|
|
for (int ch = 0; ch < s->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
|
|
set_band_parameters(s, dnch);
|
|
}
|
|
}
|
|
|
|
static int config_input(AVFilterLink *inlink)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AudioFFTDeNoiseContext *s = ctx->priv;
|
|
double wscale, sar, sum, sdiv;
|
|
int i, j, k, m, n;
|
|
|
|
s->dnch = av_calloc(inlink->channels, sizeof(*s->dnch));
|
|
if (!s->dnch)
|
|
return AVERROR(ENOMEM);
|
|
|
|
s->pts = AV_NOPTS_VALUE;
|
|
s->channels = inlink->channels;
|
|
s->sample_rate = inlink->sample_rate;
|
|
s->sample_advance = s->sample_rate / 80;
|
|
s->window_length = 3 * s->sample_advance;
|
|
s->fft_length2 = 1 << (32 - ff_clz(s->window_length));
|
|
s->fft_length = s->fft_length2 * 2;
|
|
s->buffer_length = s->fft_length * 2;
|
|
s->bin_count = s->fft_length2 + 1;
|
|
|
|
s->band_centre[0] = 80;
|
|
for (i = 1; i < 15; i++) {
|
|
s->band_centre[i] = lrint(1.5 * s->band_centre[i - 1] + 5.0);
|
|
if (s->band_centre[i] < 1000) {
|
|
s->band_centre[i] = 10 * (s->band_centre[i] / 10);
|
|
} else if (s->band_centre[i] < 5000) {
|
|
s->band_centre[i] = 50 * ((s->band_centre[i] + 20) / 50);
|
|
} else if (s->band_centre[i] < 15000) {
|
|
s->band_centre[i] = 100 * ((s->band_centre[i] + 45) / 100);
|
|
} else {
|
|
s->band_centre[i] = 1000 * ((s->band_centre[i] + 495) / 1000);
|
|
}
|
|
}
|
|
|
|
for (j = 0; j < 5; j++) {
|
|
for (k = 0; k < 5; k++) {
|
|
s->matrix_a[j + k * 5] = 0.0;
|
|
for (m = 0; m < 15; m++)
|
|
s->matrix_a[j + k * 5] += pow(m, j + k);
|
|
}
|
|
}
|
|
|
|
factor(s->matrix_a, 5);
|
|
|
|
i = 0;
|
|
for (j = 0; j < 5; j++)
|
|
for (k = 0; k < 15; k++)
|
|
s->matrix_b[i++] = pow(k, j);
|
|
|
|
i = 0;
|
|
for (j = 0; j < 15; j++)
|
|
for (k = 0; k < 5; k++)
|
|
s->matrix_c[i++] = pow(j, k);
|
|
|
|
s->window = av_calloc(s->window_length, sizeof(*s->window));
|
|
s->bin2band = av_calloc(s->bin_count, sizeof(*s->bin2band));
|
|
if (!s->window || !s->bin2band)
|
|
return AVERROR(ENOMEM);
|
|
|
|
sdiv = s->sample_rate / 17640.0;
|
|
for (i = 0; i <= s->fft_length2; i++)
|
|
s->bin2band[i] = lrint(sdiv * freq2bark((0.5 * i * s->sample_rate) / s->fft_length2));
|
|
|
|
s->number_of_bands = s->bin2band[s->fft_length2] + 1;
|
|
|
|
s->band_alpha = av_calloc(s->number_of_bands, sizeof(*s->band_alpha));
|
|
s->band_beta = av_calloc(s->number_of_bands, sizeof(*s->band_beta));
|
|
if (!s->band_alpha || !s->band_beta)
|
|
return AVERROR(ENOMEM);
|
|
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
|
|
switch (s->noise_type) {
|
|
case WHITE_NOISE:
|
|
for (i = 0; i < 15; i++)
|
|
dnch->band_noise[i] = 0;
|
|
break;
|
|
case VINYL_NOISE:
|
|
for (i = 0; i < 15; i++)
|
|
dnch->band_noise[i] = get_band_noise(s, i, 50.0, 500.5, 2125.0) + FFMAX(i - 7, 0);
|
|
break;
|
|
case SHELLAC_NOISE:
|
|
for (i = 0; i < 15; i++)
|
|
dnch->band_noise[i] = get_band_noise(s, i, 1.0, 500.0, 1.0E10) + FFMAX(i - 12, -5);
|
|
break;
|
|
case CUSTOM_NOISE:
|
|
read_custom_noise(s, ch);
|
|
break;
|
|
default:
|
|
return AVERROR_BUG;
|
|
}
|
|
|
|
|
|
dnch->sfm_threshold = 0.8;
|
|
dnch->sfm_alpha = 0.05;
|
|
for (i = 0; i < 512; i++)
|
|
dnch->sfm_fail_flags[i] = 0;
|
|
|
|
dnch->sfm_fail_total = 0;
|
|
j = FFMAX((int)(10.0 * (1.3 - dnch->sfm_threshold)), 1);
|
|
|
|
for (i = 0; i < 512; i += j) {
|
|
dnch->sfm_fail_flags[i] = 1;
|
|
dnch->sfm_fail_total += 1;
|
|
}
|
|
|
|
dnch->amt = av_calloc(s->bin_count, sizeof(*dnch->amt));
|
|
dnch->band_amt = av_calloc(s->number_of_bands, sizeof(*dnch->band_amt));
|
|
dnch->band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->band_excit));
|
|
dnch->gain = av_calloc(s->bin_count, sizeof(*dnch->gain));
|
|
dnch->prior = av_calloc(s->bin_count, sizeof(*dnch->prior));
|
|
dnch->prior_band_excit = av_calloc(s->number_of_bands, sizeof(*dnch->prior_band_excit));
|
|
dnch->clean_data = av_calloc(s->bin_count, sizeof(*dnch->clean_data));
|
|
dnch->noisy_data = av_calloc(s->bin_count, sizeof(*dnch->noisy_data));
|
|
dnch->out_samples = av_calloc(s->buffer_length, sizeof(*dnch->out_samples));
|
|
dnch->abs_var = av_calloc(s->bin_count, sizeof(*dnch->abs_var));
|
|
dnch->rel_var = av_calloc(s->bin_count, sizeof(*dnch->rel_var));
|
|
dnch->min_abs_var = av_calloc(s->bin_count, sizeof(*dnch->min_abs_var));
|
|
dnch->fft_data = av_calloc(s->fft_length2 + 1, sizeof(*dnch->fft_data));
|
|
dnch->fft = av_fft_init(av_log2(s->fft_length2), 0);
|
|
dnch->ifft = av_fft_init(av_log2(s->fft_length2), 1);
|
|
dnch->spread_function = av_calloc(s->number_of_bands * s->number_of_bands,
|
|
sizeof(*dnch->spread_function));
|
|
|
|
if (!dnch->amt ||
|
|
!dnch->band_amt ||
|
|
!dnch->band_excit ||
|
|
!dnch->gain ||
|
|
!dnch->prior ||
|
|
!dnch->prior_band_excit ||
|
|
!dnch->clean_data ||
|
|
!dnch->noisy_data ||
|
|
!dnch->out_samples ||
|
|
!dnch->fft_data ||
|
|
!dnch->abs_var ||
|
|
!dnch->rel_var ||
|
|
!dnch->min_abs_var ||
|
|
!dnch->spread_function ||
|
|
!dnch->fft ||
|
|
!dnch->ifft)
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
double *prior_band_excit = dnch->prior_band_excit;
|
|
double *prior = dnch->prior;
|
|
double min, max;
|
|
double p1, p2;
|
|
|
|
p1 = pow(0.1, 2.5 / sdiv);
|
|
p2 = pow(0.1, 1.0 / sdiv);
|
|
j = 0;
|
|
for (m = 0; m < s->number_of_bands; m++) {
|
|
for (n = 0; n < s->number_of_bands; n++) {
|
|
if (n < m) {
|
|
dnch->spread_function[j++] = pow(p2, m - n);
|
|
} else if (n > m) {
|
|
dnch->spread_function[j++] = pow(p1, n - m);
|
|
} else {
|
|
dnch->spread_function[j++] = 1.0;
|
|
}
|
|
}
|
|
}
|
|
|
|
for (m = 0; m < s->number_of_bands; m++) {
|
|
dnch->band_excit[m] = 0.0;
|
|
prior_band_excit[m] = 0.0;
|
|
}
|
|
|
|
for (m = 0; m <= s->fft_length2; m++)
|
|
dnch->band_excit[s->bin2band[m]] += 1.0;
|
|
|
|
j = 0;
|
|
for (m = 0; m < s->number_of_bands; m++) {
|
|
for (n = 0; n < s->number_of_bands; n++)
|
|
prior_band_excit[m] += dnch->spread_function[j++] * dnch->band_excit[n];
|
|
}
|
|
|
|
min = pow(0.1, 2.5);
|
|
max = pow(0.1, 1.0);
|
|
for (int i = 0; i < s->number_of_bands; i++) {
|
|
if (i < lrint(12.0 * sdiv)) {
|
|
dnch->band_excit[i] = pow(0.1, 1.45 + 0.1 * i / sdiv);
|
|
} else {
|
|
dnch->band_excit[i] = pow(0.1, 2.5 - 0.2 * (i / sdiv - 14.0));
|
|
}
|
|
dnch->band_excit[i] = av_clipd(dnch->band_excit[i], min, max);
|
|
}
|
|
|
|
for (int i = 0; i <= s->fft_length2; i++)
|
|
prior[i] = RRATIO;
|
|
for (int i = 0; i < s->buffer_length; i++)
|
|
dnch->out_samples[i] = 0;
|
|
|
|
j = 0;
|
|
for (int i = 0; i < s->number_of_bands; i++)
|
|
for (int k = 0; k < s->number_of_bands; k++)
|
|
dnch->spread_function[j++] *= dnch->band_excit[i] / prior_band_excit[i];
|
|
}
|
|
|
|
j = 0;
|
|
sar = s->sample_advance / s->sample_rate;
|
|
for (int i = 0; i <= s->fft_length2; i++) {
|
|
if ((i == s->fft_length2) || (s->bin2band[i] > j)) {
|
|
double d6 = (i - 1) * s->sample_rate / s->fft_length;
|
|
double d7 = fmin(0.008 + 2.2 / d6, 0.03);
|
|
s->band_alpha[j] = exp(-sar / d7);
|
|
s->band_beta[j] = 1.0 - s->band_alpha[j];
|
|
j = s->bin2band[i];
|
|
}
|
|
}
|
|
|
|
wscale = sqrt(16.0 / (9.0 * s->fft_length));
|
|
sum = 0.0;
|
|
for (int i = 0; i < s->window_length; i++) {
|
|
double d10 = sin(i * M_PI / s->window_length);
|
|
d10 *= wscale * d10;
|
|
s->window[i] = d10;
|
|
sum += d10 * d10;
|
|
}
|
|
|
|
s->window_weight = 0.5 * sum;
|
|
s->floor = (1LL << 48) * exp(-23.025558369790467) * s->window_weight;
|
|
s->sample_floor = s->floor * exp(4.144600506562284);
|
|
s->auto_floor = s->floor * exp(6.907667510937141);
|
|
|
|
set_parameters(s);
|
|
|
|
s->noise_band_edge[0] = FFMIN(s->fft_length2, s->fft_length * get_band_edge(s, 0) / s->sample_rate);
|
|
i = 0;
|
|
for (int j = 1; j < 16; j++) {
|
|
s->noise_band_edge[j] = FFMIN(s->fft_length2, s->fft_length * get_band_edge(s, j) / s->sample_rate);
|
|
if (s->noise_band_edge[j] > lrint(1.1 * s->noise_band_edge[j - 1]))
|
|
i++;
|
|
s->noise_band_edge[16] = i;
|
|
}
|
|
s->noise_band_count = s->noise_band_edge[16];
|
|
|
|
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->fft_length);
|
|
if (!s->fifo)
|
|
return AVERROR(ENOMEM);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void preprocess(FFTComplex *in, int len)
|
|
{
|
|
double d1, d2, d3, d4, d5, d6, d7, d8, d9, d10;
|
|
int n, i, k;
|
|
|
|
d5 = 2.0 * M_PI / len;
|
|
d8 = sin(0.5 * d5);
|
|
d8 = -2.0 * d8 * d8;
|
|
d7 = sin(d5);
|
|
d9 = 1.0 + d8;
|
|
d6 = d7;
|
|
n = len / 2;
|
|
|
|
for (i = 1; i < len / 4; i++) {
|
|
k = n - i;
|
|
d2 = 0.5 * (in[i].re + in[k].re);
|
|
d1 = 0.5 * (in[i].im - in[k].im);
|
|
d4 = 0.5 * (in[i].im + in[k].im);
|
|
d3 = 0.5 * (in[k].re - in[i].re);
|
|
in[i].re = d2 + d9 * d4 + d6 * d3;
|
|
in[i].im = d1 + d9 * d3 - d6 * d4;
|
|
in[k].re = d2 - d9 * d4 - d6 * d3;
|
|
in[k].im = -d1 + d9 * d3 - d6 * d4;
|
|
d10 = d9;
|
|
d9 += d9 * d8 - d6 * d7;
|
|
d6 += d6 * d8 + d10 * d7;
|
|
}
|
|
|
|
d2 = in[0].re;
|
|
in[0].re = d2 + in[0].im;
|
|
in[0].im = d2 - in[0].im;
|
|
}
|
|
|
|
static void postprocess(FFTComplex *in, int len)
|
|
{
|
|
double d1, d2, d3, d4, d5, d6, d7, d8, d9, d10;
|
|
int n, i, k;
|
|
|
|
d5 = 2.0 * M_PI / len;
|
|
d8 = sin(0.5 * d5);
|
|
d8 = -2.0 * d8 * d8;
|
|
d7 = sin(d5);
|
|
d9 = 1.0 + d8;
|
|
d6 = d7;
|
|
n = len / 2;
|
|
for (i = 1; i < len / 4; i++) {
|
|
k = n - i;
|
|
d2 = 0.5 * (in[i].re + in[k].re);
|
|
d1 = 0.5 * (in[i].im - in[k].im);
|
|
d4 = 0.5 * (in[i].re - in[k].re);
|
|
d3 = 0.5 * (in[i].im + in[k].im);
|
|
in[i].re = d2 - d9 * d3 - d6 * d4;
|
|
in[i].im = d1 + d9 * d4 - d6 * d3;
|
|
in[k].re = d2 + d9 * d3 + d6 * d4;
|
|
in[k].im = -d1 + d9 * d4 - d6 * d3;
|
|
d10 = d9;
|
|
d9 += d9 * d8 - d6 * d7;
|
|
d6 += d6 * d8 + d10 * d7;
|
|
}
|
|
d2 = in[0].re;
|
|
in[0].re = 0.5 * (d2 + in[0].im);
|
|
in[0].im = 0.5 * (d2 - in[0].im);
|
|
}
|
|
|
|
static void init_sample_noise(DeNoiseChannel *dnch)
|
|
{
|
|
for (int i = 0; i < 15; i++) {
|
|
dnch->noise_band_norm[i] = 0.0;
|
|
dnch->noise_band_avr[i] = 0.0;
|
|
dnch->noise_band_avi[i] = 0.0;
|
|
dnch->noise_band_var[i] = 0.0;
|
|
}
|
|
}
|
|
|
|
static void sample_noise_block(AudioFFTDeNoiseContext *s,
|
|
DeNoiseChannel *dnch,
|
|
AVFrame *in, int ch)
|
|
{
|
|
float *src = (float *)in->extended_data[ch];
|
|
double mag2, var = 0.0, avr = 0.0, avi = 0.0;
|
|
int edge, j, k, n, edgemax;
|
|
|
|
for (int i = 0; i < s->window_length; i++) {
|
|
dnch->fft_data[i].re = s->window[i] * src[i] * (1LL << 24);
|
|
dnch->fft_data[i].im = 0.0;
|
|
}
|
|
|
|
for (int i = s->window_length; i < s->fft_length2; i++) {
|
|
dnch->fft_data[i].re = 0.0;
|
|
dnch->fft_data[i].im = 0.0;
|
|
}
|
|
|
|
av_fft_permute(dnch->fft, dnch->fft_data);
|
|
av_fft_calc(dnch->fft, dnch->fft_data);
|
|
|
|
preprocess(dnch->fft_data, s->fft_length);
|
|
|
|
edge = s->noise_band_edge[0];
|
|
j = edge;
|
|
k = 0;
|
|
n = j;
|
|
edgemax = fmin(s->fft_length2, s->noise_band_edge[15]);
|
|
dnch->fft_data[s->fft_length2].re = dnch->fft_data[0].im;
|
|
dnch->fft_data[0].im = 0.0;
|
|
dnch->fft_data[s->fft_length2].im = 0.0;
|
|
|
|
for (int i = j; i <= edgemax; i++) {
|
|
if ((i == j) && (i < edgemax)) {
|
|
if (j > edge) {
|
|
dnch->noise_band_norm[k - 1] += j - edge;
|
|
dnch->noise_band_avr[k - 1] += avr;
|
|
dnch->noise_band_avi[k - 1] += avi;
|
|
dnch->noise_band_var[k - 1] += var;
|
|
}
|
|
k++;
|
|
edge = j;
|
|
j = s->noise_band_edge[k];
|
|
if (k == 15) {
|
|
j++;
|
|
}
|
|
var = 0.0;
|
|
avr = 0.0;
|
|
avi = 0.0;
|
|
}
|
|
avr += dnch->fft_data[n].re;
|
|
avi += dnch->fft_data[n].im;
|
|
mag2 = dnch->fft_data[n].re * dnch->fft_data[n].re +
|
|
dnch->fft_data[n].im * dnch->fft_data[n].im;
|
|
|
|
mag2 = fmax(mag2, s->sample_floor);
|
|
|
|
dnch->noisy_data[i] = mag2;
|
|
var += mag2;
|
|
n++;
|
|
}
|
|
|
|
dnch->noise_band_norm[k - 1] += j - edge;
|
|
dnch->noise_band_avr[k - 1] += avr;
|
|
dnch->noise_band_avi[k - 1] += avi;
|
|
dnch->noise_band_var[k - 1] += var;
|
|
}
|
|
|
|
static void finish_sample_noise(AudioFFTDeNoiseContext *s,
|
|
DeNoiseChannel *dnch,
|
|
double *sample_noise)
|
|
{
|
|
for (int i = 0; i < s->noise_band_count; i++) {
|
|
dnch->noise_band_avr[i] /= dnch->noise_band_norm[i];
|
|
dnch->noise_band_avi[i] /= dnch->noise_band_norm[i];
|
|
dnch->noise_band_var[i] /= dnch->noise_band_norm[i];
|
|
dnch->noise_band_var[i] -= dnch->noise_band_avr[i] * dnch->noise_band_avr[i] +
|
|
dnch->noise_band_avi[i] * dnch->noise_band_avi[i];
|
|
dnch->noise_band_auto_var[i] = dnch->noise_band_var[i];
|
|
sample_noise[i] = (1.0 / C) * log(dnch->noise_band_var[i] / s->floor) - 100.0;
|
|
}
|
|
if (s->noise_band_count < 15) {
|
|
for (int i = s->noise_band_count; i < 15; i++)
|
|
sample_noise[i] = sample_noise[i - 1];
|
|
}
|
|
}
|
|
|
|
static void set_noise_profile(AudioFFTDeNoiseContext *s,
|
|
DeNoiseChannel *dnch,
|
|
double *sample_noise,
|
|
int new_profile)
|
|
{
|
|
int new_band_noise[15];
|
|
double temp[15];
|
|
double sum = 0.0, d1;
|
|
float new_noise_floor;
|
|
int i, n;
|
|
|
|
for (int m = 0; m < 15; m++)
|
|
temp[m] = sample_noise[m];
|
|
|
|
if (new_profile) {
|
|
i = 0;
|
|
for (int m = 0; m < 5; m++) {
|
|
sum = 0.0;
|
|
for (n = 0; n < 15; n++)
|
|
sum += s->matrix_b[i++] * temp[n];
|
|
s->vector_b[m] = sum;
|
|
}
|
|
solve(s->matrix_a, s->vector_b, 5);
|
|
i = 0;
|
|
for (int m = 0; m < 15; m++) {
|
|
sum = 0.0;
|
|
for (n = 0; n < 5; n++)
|
|
sum += s->matrix_c[i++] * s->vector_b[n];
|
|
temp[m] = sum;
|
|
}
|
|
}
|
|
|
|
sum = 0.0;
|
|
for (int m = 0; m < 15; m++)
|
|
sum += temp[m];
|
|
|
|
d1 = (int)(sum / 15.0 - 0.5);
|
|
if (!new_profile)
|
|
i = lrint(temp[7] - d1);
|
|
|
|
for (d1 -= dnch->band_noise[7] - i; d1 > -20.0; d1 -= 1.0)
|
|
;
|
|
|
|
for (int m = 0; m < 15; m++)
|
|
temp[m] -= d1;
|
|
|
|
new_noise_floor = d1 + 2.5;
|
|
|
|
if (new_profile) {
|
|
av_log(s, AV_LOG_INFO, "bn=");
|
|
for (int m = 0; m < 15; m++) {
|
|
new_band_noise[m] = lrint(temp[m]);
|
|
new_band_noise[m] = av_clip(new_band_noise[m], -24, 24);
|
|
av_log(s, AV_LOG_INFO, "%d ", new_band_noise[m]);
|
|
}
|
|
av_log(s, AV_LOG_INFO, "\n");
|
|
memcpy(dnch->band_noise, new_band_noise, sizeof(new_band_noise));
|
|
}
|
|
|
|
if (s->track_noise)
|
|
s->noise_floor = new_noise_floor;
|
|
}
|
|
|
|
typedef struct ThreadData {
|
|
AVFrame *in;
|
|
} ThreadData;
|
|
|
|
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
|
|
{
|
|
AudioFFTDeNoiseContext *s = ctx->priv;
|
|
ThreadData *td = arg;
|
|
AVFrame *in = td->in;
|
|
const int start = (in->channels * jobnr) / nb_jobs;
|
|
const int end = (in->channels * (jobnr+1)) / nb_jobs;
|
|
|
|
for (int ch = start; ch < end; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
const float *src = (const float *)in->extended_data[ch];
|
|
double *dst = dnch->out_samples;
|
|
|
|
if (s->track_noise) {
|
|
int i = s->block_count & 0x1FF;
|
|
|
|
if (dnch->sfm_fail_flags[i])
|
|
dnch->sfm_fail_total--;
|
|
dnch->sfm_fail_flags[i] = 0;
|
|
dnch->sfm_threshold *= 1.0 - dnch->sfm_alpha;
|
|
dnch->sfm_threshold += dnch->sfm_alpha * (0.5 + (1.0 / 640) * dnch->sfm_fail_total);
|
|
}
|
|
|
|
for (int m = 0; m < s->window_length; m++) {
|
|
dnch->fft_data[m].re = s->window[m] * src[m] * (1LL << 24);
|
|
dnch->fft_data[m].im = 0;
|
|
}
|
|
|
|
for (int m = s->window_length; m < s->fft_length2; m++) {
|
|
dnch->fft_data[m].re = 0;
|
|
dnch->fft_data[m].im = 0;
|
|
}
|
|
|
|
av_fft_permute(dnch->fft, dnch->fft_data);
|
|
av_fft_calc(dnch->fft, dnch->fft_data);
|
|
|
|
preprocess(dnch->fft_data, s->fft_length);
|
|
process_frame(s, dnch, dnch->fft_data,
|
|
dnch->prior,
|
|
dnch->prior_band_excit,
|
|
s->track_noise);
|
|
postprocess(dnch->fft_data, s->fft_length);
|
|
|
|
av_fft_permute(dnch->ifft, dnch->fft_data);
|
|
av_fft_calc(dnch->ifft, dnch->fft_data);
|
|
|
|
for (int m = 0; m < s->window_length; m++)
|
|
dst[m] += s->window[m] * dnch->fft_data[m].re / (1LL << 24);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void get_auto_noise_levels(AudioFFTDeNoiseContext *s,
|
|
DeNoiseChannel *dnch,
|
|
double *levels)
|
|
{
|
|
if (s->noise_band_count > 0) {
|
|
for (int i = 0; i < s->noise_band_count; i++) {
|
|
levels[i] = (1.0 / C) * log(dnch->noise_band_auto_var[i] / s->floor) - 100.0;
|
|
}
|
|
if (s->noise_band_count < 15) {
|
|
for (int i = s->noise_band_count; i < 15; i++)
|
|
levels[i] = levels[i - 1];
|
|
}
|
|
} else {
|
|
for (int i = 0; i < 15; i++) {
|
|
levels[i] = -100.0;
|
|
}
|
|
}
|
|
}
|
|
|
|
static int output_frame(AVFilterLink *inlink)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
AudioFFTDeNoiseContext *s = ctx->priv;
|
|
AVFrame *out = NULL, *in = NULL;
|
|
ThreadData td;
|
|
int ret = 0;
|
|
|
|
in = ff_get_audio_buffer(outlink, s->window_length);
|
|
if (!in)
|
|
return AVERROR(ENOMEM);
|
|
|
|
ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length);
|
|
if (ret < 0)
|
|
goto end;
|
|
|
|
if (s->track_noise) {
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
double levels[15];
|
|
|
|
get_auto_noise_levels(s, dnch, levels);
|
|
set_noise_profile(s, dnch, levels, 0);
|
|
}
|
|
|
|
if (s->noise_floor != s->last_noise_floor)
|
|
set_parameters(s);
|
|
}
|
|
|
|
if (s->sample_noise_start) {
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
|
|
init_sample_noise(dnch);
|
|
}
|
|
s->sample_noise_start = 0;
|
|
s->sample_noise = 1;
|
|
}
|
|
|
|
if (s->sample_noise) {
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
|
|
sample_noise_block(s, dnch, in, ch);
|
|
}
|
|
}
|
|
|
|
if (s->sample_noise_end) {
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
double sample_noise[15];
|
|
|
|
finish_sample_noise(s, dnch, sample_noise);
|
|
set_noise_profile(s, dnch, sample_noise, 1);
|
|
set_band_parameters(s, dnch);
|
|
}
|
|
s->sample_noise = 0;
|
|
s->sample_noise_end = 0;
|
|
}
|
|
|
|
s->block_count++;
|
|
td.in = in;
|
|
ctx->internal->execute(ctx, filter_channel, &td, NULL,
|
|
FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
|
|
|
|
out = ff_get_audio_buffer(outlink, s->sample_advance);
|
|
if (!out) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto end;
|
|
}
|
|
|
|
for (int ch = 0; ch < inlink->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
double *src = dnch->out_samples;
|
|
float *orig = (float *)in->extended_data[ch];
|
|
float *dst = (float *)out->extended_data[ch];
|
|
|
|
switch (s->output_mode) {
|
|
case IN_MODE:
|
|
for (int m = 0; m < s->sample_advance; m++)
|
|
dst[m] = orig[m];
|
|
break;
|
|
case OUT_MODE:
|
|
for (int m = 0; m < s->sample_advance; m++)
|
|
dst[m] = src[m];
|
|
break;
|
|
case NOISE_MODE:
|
|
for (int m = 0; m < s->sample_advance; m++)
|
|
dst[m] = orig[m] - src[m];
|
|
break;
|
|
default:
|
|
av_frame_free(&out);
|
|
ret = AVERROR_BUG;
|
|
goto end;
|
|
}
|
|
memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src));
|
|
memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
|
|
}
|
|
|
|
av_audio_fifo_drain(s->fifo, s->sample_advance);
|
|
|
|
out->pts = s->pts;
|
|
ret = ff_filter_frame(outlink, out);
|
|
if (ret < 0)
|
|
goto end;
|
|
s->pts += av_rescale_q(s->sample_advance, (AVRational){1, outlink->sample_rate}, outlink->time_base);
|
|
end:
|
|
av_frame_free(&in);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int activate(AVFilterContext *ctx)
|
|
{
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
AudioFFTDeNoiseContext *s = ctx->priv;
|
|
AVFrame *frame = NULL;
|
|
int ret;
|
|
|
|
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
|
|
|
|
ret = ff_inlink_consume_frame(inlink, &frame);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (ret > 0) {
|
|
if (s->pts == AV_NOPTS_VALUE)
|
|
s->pts = frame->pts;
|
|
|
|
ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
|
|
av_frame_free(&frame);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
if (av_audio_fifo_size(s->fifo) >= s->window_length)
|
|
return output_frame(inlink);
|
|
|
|
FF_FILTER_FORWARD_STATUS(inlink, outlink);
|
|
if (ff_outlink_frame_wanted(outlink) &&
|
|
av_audio_fifo_size(s->fifo) < s->window_length) {
|
|
ff_inlink_request_frame(inlink);
|
|
return 0;
|
|
}
|
|
|
|
return FFERROR_NOT_READY;
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
AudioFFTDeNoiseContext *s = ctx->priv;
|
|
|
|
av_freep(&s->window);
|
|
av_freep(&s->bin2band);
|
|
av_freep(&s->band_alpha);
|
|
av_freep(&s->band_beta);
|
|
|
|
if (s->dnch) {
|
|
for (int ch = 0; ch < s->channels; ch++) {
|
|
DeNoiseChannel *dnch = &s->dnch[ch];
|
|
av_freep(&dnch->amt);
|
|
av_freep(&dnch->band_amt);
|
|
av_freep(&dnch->band_excit);
|
|
av_freep(&dnch->gain);
|
|
av_freep(&dnch->prior);
|
|
av_freep(&dnch->prior_band_excit);
|
|
av_freep(&dnch->clean_data);
|
|
av_freep(&dnch->noisy_data);
|
|
av_freep(&dnch->out_samples);
|
|
av_freep(&dnch->spread_function);
|
|
av_freep(&dnch->abs_var);
|
|
av_freep(&dnch->rel_var);
|
|
av_freep(&dnch->min_abs_var);
|
|
av_freep(&dnch->fft_data);
|
|
av_fft_end(dnch->fft);
|
|
dnch->fft = NULL;
|
|
av_fft_end(dnch->ifft);
|
|
dnch->ifft = NULL;
|
|
}
|
|
av_freep(&s->dnch);
|
|
}
|
|
|
|
av_audio_fifo_free(s->fifo);
|
|
}
|
|
|
|
static int query_formats(AVFilterContext *ctx)
|
|
{
|
|
AVFilterFormats *formats = NULL;
|
|
AVFilterChannelLayouts *layouts = NULL;
|
|
static const enum AVSampleFormat sample_fmts[] = {
|
|
AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE
|
|
};
|
|
int 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;
|
|
|
|
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_all_samplerates();
|
|
return ff_set_common_samplerates(ctx, formats);
|
|
}
|
|
|
|
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
|
|
char *res, int res_len, int flags)
|
|
{
|
|
AudioFFTDeNoiseContext *s = ctx->priv;
|
|
int need_reset = 0;
|
|
int ret = 0;
|
|
|
|
if (!strcmp(cmd, "sample_noise") ||
|
|
!strcmp(cmd, "sn")) {
|
|
if (!strcmp(args, "start")) {
|
|
s->sample_noise_start = 1;
|
|
s->sample_noise_end = 0;
|
|
} else if (!strcmp(args, "end") ||
|
|
!strcmp(args, "stop")) {
|
|
s->sample_noise_start = 0;
|
|
s->sample_noise_end = 1;
|
|
}
|
|
} else {
|
|
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
|
|
if (ret < 0)
|
|
return ret;
|
|
need_reset = 1;
|
|
}
|
|
|
|
if (need_reset)
|
|
set_parameters(s);
|
|
|
|
return 0;
|
|
}
|
|
|
|
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 }
|
|
};
|
|
|
|
AVFilter ff_af_afftdn = {
|
|
.name = "afftdn",
|
|
.description = NULL_IF_CONFIG_SMALL("Denoise audio samples using FFT."),
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(AudioFFTDeNoiseContext),
|
|
.priv_class = &afftdn_class,
|
|
.activate = activate,
|
|
.uninit = uninit,
|
|
.inputs = inputs,
|
|
.outputs = outputs,
|
|
.process_command = process_command,
|
|
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
|
|
AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|