mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
3 years ago
parent
e6f0cec880
commit
163e737c17
5 changed files with 623 additions and 415 deletions
@ -0,0 +1,99 @@ |
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/*
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* Copyright (c) 2017 Paul B Mahol |
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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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#ifndef AVFILTER_AFIR_H |
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#define AVFILTER_AFIR_H |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/frame.h" |
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#include "libavutil/rational.h" |
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#include "libavutil/tx.h" |
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#include "avfilter.h" |
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#include "af_afirdsp.h" |
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typedef struct AudioFIRSegment { |
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int nb_partitions; |
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int part_size; |
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int block_size; |
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int fft_length; |
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int coeff_size; |
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int input_size; |
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int input_offset; |
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int *output_offset; |
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int *part_index; |
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AVFrame *sumin; |
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AVFrame *sumout; |
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AVFrame *blockin; |
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AVFrame *blockout; |
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AVFrame *buffer; |
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AVFrame *coeff; |
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AVFrame *input; |
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AVFrame *output; |
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AVTXContext **tx, **itx; |
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av_tx_fn tx_fn, itx_fn; |
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} AudioFIRSegment; |
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typedef struct AudioFIRContext { |
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const AVClass *class; |
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float wet_gain; |
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float dry_gain; |
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float length; |
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int gtype; |
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float ir_gain; |
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int ir_format; |
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float max_ir_len; |
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int response; |
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int w, h; |
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AVRational frame_rate; |
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int ir_channel; |
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int minp; |
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int maxp; |
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int nb_irs; |
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int selir; |
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int precision; |
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int format; |
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double gain; |
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int eof_coeffs[32]; |
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int have_coeffs; |
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int nb_taps; |
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int nb_channels; |
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int nb_coef_channels; |
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int one2many; |
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AudioFIRSegment seg[1024]; |
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int nb_segments; |
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AVFrame *in; |
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AVFrame *ir[32]; |
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AVFrame *video; |
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int min_part_size; |
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int64_t pts; |
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AudioFIRDSPContext afirdsp; |
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AVFloatDSPContext *fdsp; |
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} AudioFIRContext; |
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#endif /* AVFILTER_AFIR_H */ |
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@ -0,0 +1,392 @@ |
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/*
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* Copyright (c) 2017 Paul B Mahol |
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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 "avfilter.h" |
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#include "formats.h" |
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#include "internal.h" |
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#include "audio.h" |
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#undef ctype |
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#undef ftype |
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#undef SQRT |
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#undef SAMPLE_FORMAT |
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#if DEPTH == 32 |
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#define SAMPLE_FORMAT float |
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#define SQRT sqrtf |
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#define ctype AVComplexFloat |
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#define ftype float |
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#else |
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#define SAMPLE_FORMAT double |
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#define SQRT sqrt |
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#define ctype AVComplexDouble |
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#define ftype double |
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#endif |
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#define fn3(a,b) a##_##b |
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#define fn2(a,b) fn3(a,b) |
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#define fn(a) fn2(a, SAMPLE_FORMAT) |
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static void fn(draw_response)(AVFilterContext *ctx, AVFrame *out) |
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{ |
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AudioFIRContext *s = ctx->priv; |
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ftype *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN; |
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ftype min_delay = FLT_MAX, max_delay = FLT_MIN; |
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int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1; |
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char text[32]; |
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int channel, i, x; |
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memset(out->data[0], 0, s->h * out->linesize[0]); |
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phase = av_malloc_array(s->w, sizeof(*phase)); |
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mag = av_malloc_array(s->w, sizeof(*mag)); |
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delay = av_malloc_array(s->w, sizeof(*delay)); |
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if (!mag || !phase || !delay) |
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goto end; |
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channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1); |
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for (i = 0; i < s->w; i++) { |
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const ftype *src = (const ftype *)s->ir[s->selir]->extended_data[channel]; |
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double w = i * M_PI / (s->w - 1); |
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double div, real_num = 0., imag_num = 0., real = 0., imag = 0.; |
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for (x = 0; x < s->nb_taps; x++) { |
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real += cos(-x * w) * src[x]; |
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imag += sin(-x * w) * src[x]; |
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real_num += cos(-x * w) * src[x] * x; |
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imag_num += sin(-x * w) * src[x] * x; |
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} |
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mag[i] = hypot(real, imag); |
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phase[i] = atan2(imag, real); |
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div = real * real + imag * imag; |
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delay[i] = (real_num * real + imag_num * imag) / div; |
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min = fminf(min, mag[i]); |
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max = fmaxf(max, mag[i]); |
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min_delay = fminf(min_delay, delay[i]); |
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max_delay = fmaxf(max_delay, delay[i]); |
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} |
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for (i = 0; i < s->w; i++) { |
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int ymag = mag[i] / max * (s->h - 1); |
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int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1); |
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int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1); |
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ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1); |
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yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1); |
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ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1); |
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if (prev_ymag < 0) |
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prev_ymag = ymag; |
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if (prev_yphase < 0) |
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prev_yphase = yphase; |
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if (prev_ydelay < 0) |
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prev_ydelay = ydelay; |
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draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF); |
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draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00); |
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draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF); |
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prev_ymag = ymag; |
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prev_yphase = yphase; |
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prev_ydelay = ydelay; |
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} |
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if (s->w > 400 && s->h > 100) { |
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drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD); |
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snprintf(text, sizeof(text), "%.2f", max); |
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drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD); |
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drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD); |
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snprintf(text, sizeof(text), "%.2f", min); |
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drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD); |
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drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD); |
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snprintf(text, sizeof(text), "%.2f", max_delay); |
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drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD); |
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drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD); |
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snprintf(text, sizeof(text), "%.2f", min_delay); |
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drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD); |
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} |
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end: |
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av_free(delay); |
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av_free(phase); |
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av_free(mag); |
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} |
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static void fn(convert_channels)(AVFilterContext *ctx, AudioFIRContext *s) |
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{ |
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for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) { |
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch]; |
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int toffset = 0; |
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for (int i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) |
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time[i] = 0; |
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av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch); |
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for (int segment = 0; segment < s->nb_segments; segment++) { |
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AudioFIRSegment *seg = &s->seg[segment]; |
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ftype *blockin = (ftype *)seg->blockin->extended_data[ch]; |
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ftype *blockout = (ftype *)seg->blockout->extended_data[ch]; |
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ctype *coeff = (ctype *)seg->coeff->extended_data[ch]; |
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av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment); |
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for (int i = 0; i < seg->nb_partitions; i++) { |
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const int coffset = i * seg->coeff_size; |
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const int remaining = s->nb_taps - toffset; |
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const int size = remaining >= seg->part_size ? seg->part_size : remaining; |
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if (size < 8) { |
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for (int n = 0; n < size; n++) |
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coeff[coffset + n].re = time[toffset + n]; |
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toffset += size; |
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continue; |
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} |
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memset(blockin, 0, sizeof(*blockin) * seg->fft_length); |
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memcpy(blockin, time + toffset, size * sizeof(*blockin)); |
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seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(ftype)); |
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for (int n = 0; n < seg->part_size + 1; n++) { |
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coeff[coffset + n].re = blockout[2 * n]; |
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coeff[coffset + n].im = blockout[2 * n + 1]; |
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} |
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toffset += size; |
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} |
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av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions); |
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av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size); |
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av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size); |
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av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length); |
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av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size); |
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av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size); |
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av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset); |
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} |
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} |
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} |
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static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps) |
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{ |
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ftype power = 0; |
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int ch; |
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switch (s->gtype) { |
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case -1: |
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/* nothing to do */ |
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break; |
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case 0: |
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for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) { |
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch]; |
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for (int i = 0; i < cur_nb_taps; i++) |
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power += FFABS(time[i]); |
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} |
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s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power; |
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break; |
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case 1: |
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for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) { |
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch]; |
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for (int i = 0; i < cur_nb_taps; i++) |
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power += time[i]; |
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} |
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s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power; |
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break; |
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case 2: |
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for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) { |
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch]; |
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for (int i = 0; i < cur_nb_taps; i++) |
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power += time[i] * time[i]; |
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} |
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s->gain = SQRT(ch / power); |
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break; |
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default: |
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return AVERROR_BUG; |
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} |
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s->gain = FFMIN(s->gain * s->ir_gain, 1.); |
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av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain); |
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for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) { |
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ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch]; |
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#if DEPTH == 32 |
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s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4)); |
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#else |
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s->fdsp->vector_dmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 8)); |
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#endif |
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} |
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return 0; |
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} |
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static void fn(direct)(const ftype *in, const ctype *ir, int len, ftype *out) |
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{ |
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for (int n = 0; n < len; n++) |
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for (int m = 0; m <= n; m++) |
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out[n] += ir[m].re * in[n - m]; |
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} |
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static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples) |
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{ |
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if ((nb_samples & 15) == 0 && nb_samples >= 16) { |
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#if DEPTH == 32 |
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s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples); |
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#else |
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s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples); |
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#endif |
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} else { |
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for (int n = 0; n < nb_samples; n++) |
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dst[n] += src[n]; |
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} |
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} |
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static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offset) |
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{ |
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AudioFIRContext *s = ctx->priv; |
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const ftype *in = (const ftype *)s->in->extended_data[ch] + offset; |
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ftype *blockin, *blockout, *buf, *ptr = (ftype *)out->extended_data[ch] + offset; |
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const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset); |
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int n, i, j; |
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for (int segment = 0; segment < s->nb_segments; segment++) { |
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AudioFIRSegment *seg = &s->seg[segment]; |
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ftype *src = (ftype *)seg->input->extended_data[ch]; |
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ftype *dst = (ftype *)seg->output->extended_data[ch]; |
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ftype *sumin = (ftype *)seg->sumin->extended_data[ch]; |
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ftype *sumout = (ftype *)seg->sumout->extended_data[ch]; |
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if (s->min_part_size >= 8) { |
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#if DEPTH == 32 |
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s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4)); |
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#else |
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s->fdsp->vector_dmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 8)); |
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#endif |
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emms_c(); |
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} else { |
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for (n = 0; n < nb_samples; n++) |
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src[seg->input_offset + n] = in[n] * s->dry_gain; |
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} |
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seg->output_offset[ch] += s->min_part_size; |
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if (seg->output_offset[ch] == seg->part_size) { |
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seg->output_offset[ch] = 0; |
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} else { |
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memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src)); |
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dst += seg->output_offset[ch]; |
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fn(fir_fadd)(s, ptr, dst, nb_samples); |
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continue; |
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} |
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if (seg->part_size < 8) { |
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memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions); |
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j = seg->part_index[ch]; |
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for (i = 0; i < seg->nb_partitions; i++) { |
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const int coffset = j * seg->coeff_size; |
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const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset; |
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fn(direct)(src, coeff, nb_samples, dst); |
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if (j == 0) |
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j = seg->nb_partitions; |
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j--; |
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} |
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seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions; |
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memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src)); |
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for (n = 0; n < nb_samples; n++) { |
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ptr[n] += dst[n]; |
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} |
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continue; |
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} |
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memset(sumin, 0, sizeof(*sumin) * seg->fft_length); |
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blockin = (ftype *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size; |
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blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size; |
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memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size)); |
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memcpy(blockin, src, sizeof(*src) * seg->part_size); |
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seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(ftype)); |
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j = seg->part_index[ch]; |
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for (i = 0; i < seg->nb_partitions; i++) { |
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const int coffset = j * seg->coeff_size; |
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const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + i * seg->block_size; |
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const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset; |
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#if DEPTH == 32 |
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s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size); |
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#else |
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s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size); |
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#endif |
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if (j == 0) |
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j = seg->nb_partitions; |
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j--; |
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} |
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seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ftype)); |
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buf = (ftype *)seg->buffer->extended_data[ch]; |
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fn(fir_fadd)(s, buf, sumout, seg->part_size); |
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memcpy(dst, buf, seg->part_size * sizeof(*dst)); |
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buf = (ftype *)seg->buffer->extended_data[ch]; |
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memcpy(buf, sumout + seg->part_size, seg->part_size * sizeof(*buf)); |
||||
|
||||
seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions; |
||||
|
||||
memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src)); |
||||
|
||||
fn(fir_fadd)(s, ptr, dst, nb_samples); |
||||
} |
||||
|
||||
if (s->min_part_size >= 8) { |
||||
#if DEPTH == 32 |
||||
s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4)); |
||||
#else |
||||
s->fdsp->vector_dmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 8)); |
||||
#endif |
||||
emms_c(); |
||||
} else { |
||||
for (n = 0; n < nb_samples; n++) |
||||
ptr[n] *= s->wet_gain; |
||||
} |
||||
|
||||
return 0; |
||||
} |
||||
|
||||
|
||||
Loading…
Reference in new issue