mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
2 years ago
parent
f247a3d82d
commit
a2f4adf680
6 changed files with 395 additions and 1 deletions
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/*
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* Copyright (c) 2023 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 "libavutil/common.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/opt.h" |
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#include "audio.h" |
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#include "avfilter.h" |
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#include "formats.h" |
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#include "filters.h" |
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#include "internal.h" |
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enum OutModes { |
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IN_MODE, |
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DESIRED_MODE, |
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OUT_MODE, |
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NOISE_MODE, |
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NB_OMODES |
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}; |
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typedef struct AudioRLSContext { |
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const AVClass *class; |
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int order; |
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float lambda; |
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float delta; |
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int output_mode; |
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int kernel_size; |
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AVFrame *offset; |
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AVFrame *delay; |
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AVFrame *coeffs; |
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AVFrame *p, *dp; |
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AVFrame *gains; |
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AVFrame *u, *tmp; |
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AVFrame *frame[2]; |
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AVFloatDSPContext *fdsp; |
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} AudioRLSContext; |
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#define OFFSET(x) offsetof(AudioRLSContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM |
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static const AVOption arls_options[] = { |
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{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A }, |
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{ "lambda", "set the filter lambda", OFFSET(lambda), AV_OPT_TYPE_FLOAT, {.dbl=1.f}, 0, 1, AT }, |
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{ "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=2.f}, 0, INT16_MAX, A }, |
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{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" }, |
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{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" }, |
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{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" }, |
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{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" }, |
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{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(arls); |
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static float fir_sample(AudioRLSContext *s, float sample, float *delay, |
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float *coeffs, float *tmp, int *offset) |
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{ |
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const int order = s->order; |
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float output; |
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delay[*offset] = sample; |
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memcpy(tmp, coeffs + order - *offset, order * sizeof(float)); |
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output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size); |
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if (--(*offset) < 0) |
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*offset = order - 1; |
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return output; |
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} |
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static float process_sample(AudioRLSContext *s, float input, float desired, int ch) |
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{ |
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float *coeffs = (float *)s->coeffs->extended_data[ch]; |
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float *delay = (float *)s->delay->extended_data[ch]; |
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float *gains = (float *)s->gains->extended_data[ch]; |
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float *tmp = (float *)s->tmp->extended_data[ch]; |
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float *u = (float *)s->u->extended_data[ch]; |
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float *p = (float *)s->p->extended_data[ch]; |
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float *dp = (float *)s->dp->extended_data[ch]; |
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int *offsetp = (int *)s->offset->extended_data[ch]; |
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const int kernel_size = s->kernel_size; |
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const int order = s->order; |
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const float lambda = s->lambda; |
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int offset = *offsetp; |
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float g = lambda; |
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float output, e; |
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delay[offset + order] = input; |
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output = fir_sample(s, input, delay, coeffs, tmp, offsetp); |
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e = desired - output; |
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for (int i = 0, pos = offset; i < order; i++, pos++) { |
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const int ikernel_size = i * kernel_size; |
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u[i] = 0.f; |
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for (int k = 0, pos = offset; k < order; k++, pos++) |
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u[i] += p[ikernel_size + k] * delay[pos]; |
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g += u[i] * delay[pos]; |
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} |
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g = 1.f / g; |
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for (int i = 0; i < order; i++) { |
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const int ikernel_size = i * kernel_size; |
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gains[i] = u[i] * g; |
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coeffs[i] = coeffs[order + i] = coeffs[i] + gains[i] * e; |
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tmp[i] = 0.f; |
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for (int k = 0, pos = offset; k < order; k++, pos++) |
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tmp[i] += p[ikernel_size + k] * delay[pos]; |
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} |
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for (int i = 0; i < order; i++) { |
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const int ikernel_size = i * kernel_size; |
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for (int k = 0; k < order; k++) |
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dp[ikernel_size + k] = gains[i] * tmp[k]; |
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} |
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for (int i = 0; i < order; i++) { |
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const int ikernel_size = i * kernel_size; |
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for (int k = 0; k < order; k++) |
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p[ikernel_size + k] = (p[ikernel_size + k] - (dp[ikernel_size + k] + dp[kernel_size * k + i]) * 0.5f) * lambda; |
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} |
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switch (s->output_mode) { |
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case IN_MODE: output = input; break; |
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case DESIRED_MODE: output = desired; break; |
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case OUT_MODE: output = desired - output; break; |
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case NOISE_MODE: output = input - output; break; |
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} |
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return output; |
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} |
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static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
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{ |
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AudioRLSContext *s = ctx->priv; |
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AVFrame *out = arg; |
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const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs; |
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const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; |
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for (int c = start; c < end; c++) { |
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const float *input = (const float *)s->frame[0]->extended_data[c]; |
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const float *desired = (const float *)s->frame[1]->extended_data[c]; |
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float *output = (float *)out->extended_data[c]; |
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for (int n = 0; n < out->nb_samples; n++) { |
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output[n] = process_sample(s, input[n], desired[n], c); |
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if (ctx->is_disabled) |
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output[n] = input[n]; |
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} |
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} |
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return 0; |
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} |
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static int activate(AVFilterContext *ctx) |
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{ |
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AudioRLSContext *s = ctx->priv; |
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int i, ret, status; |
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int nb_samples; |
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int64_t pts; |
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); |
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nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), |
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ff_inlink_queued_samples(ctx->inputs[1])); |
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for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) { |
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if (s->frame[i]) |
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continue; |
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if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) { |
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ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]); |
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if (ret < 0) |
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return ret; |
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} |
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} |
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if (s->frame[0] && s->frame[1]) { |
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AVFrame *out; |
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out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples); |
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if (!out) { |
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av_frame_free(&s->frame[0]); |
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av_frame_free(&s->frame[1]); |
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return AVERROR(ENOMEM); |
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} |
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ff_filter_execute(ctx, process_channels, out, NULL, |
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FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); |
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out->pts = s->frame[0]->pts; |
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av_frame_free(&s->frame[0]); |
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av_frame_free(&s->frame[1]); |
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ret = ff_filter_frame(ctx->outputs[0], out); |
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if (ret < 0) |
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return ret; |
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} |
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if (!nb_samples) { |
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for (i = 0; i < 2; i++) { |
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { |
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ff_outlink_set_status(ctx->outputs[0], status, pts); |
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return 0; |
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} |
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} |
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} |
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if (ff_outlink_frame_wanted(ctx->outputs[0])) { |
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for (i = 0; i < 2; i++) { |
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if (ff_inlink_queued_samples(ctx->inputs[i]) > 0) |
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continue; |
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ff_inlink_request_frame(ctx->inputs[i]); |
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return 0; |
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} |
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} |
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return 0; |
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} |
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static int config_output(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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AudioRLSContext *s = ctx->priv; |
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s->kernel_size = FFALIGN(s->order, 16); |
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if (!s->offset) |
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s->offset = ff_get_audio_buffer(outlink, 1); |
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if (!s->delay) |
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s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size); |
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if (!s->coeffs) |
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s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size); |
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if (!s->gains) |
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s->gains = ff_get_audio_buffer(outlink, s->kernel_size); |
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if (!s->p) |
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s->p = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size); |
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if (!s->dp) |
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s->dp = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size); |
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if (!s->u) |
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s->u = ff_get_audio_buffer(outlink, s->kernel_size); |
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if (!s->tmp) |
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s->tmp = ff_get_audio_buffer(outlink, s->kernel_size); |
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if (!s->delay || !s->coeffs || !s->p || !s->dp || !s->gains || !s->offset || !s->u || !s->tmp) |
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return AVERROR(ENOMEM); |
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for (int ch = 0; ch < s->offset->ch_layout.nb_channels; ch++) { |
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int *dst = (int *)s->offset->extended_data[ch]; |
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for (int i = 0; i < s->kernel_size; i++) |
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dst[0] = s->kernel_size - 1; |
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} |
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for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) { |
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float *dst = (float *)s->p->extended_data[ch]; |
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for (int i = 0; i < s->kernel_size; i++) |
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dst[i * s->kernel_size + i] = s->delta; |
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} |
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return 0; |
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} |
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static av_cold int init(AVFilterContext *ctx) |
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{ |
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AudioRLSContext *s = ctx->priv; |
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s->fdsp = avpriv_float_dsp_alloc(0); |
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if (!s->fdsp) |
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return AVERROR(ENOMEM); |
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return 0; |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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AudioRLSContext *s = ctx->priv; |
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av_freep(&s->fdsp); |
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av_frame_free(&s->delay); |
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av_frame_free(&s->coeffs); |
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av_frame_free(&s->gains); |
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av_frame_free(&s->offset); |
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av_frame_free(&s->p); |
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av_frame_free(&s->dp); |
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av_frame_free(&s->u); |
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av_frame_free(&s->tmp); |
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} |
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static const AVFilterPad inputs[] = { |
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{ |
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.name = "input", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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{ |
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.name = "desired", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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}; |
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static const AVFilterPad outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.config_props = config_output, |
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}, |
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}; |
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const AVFilter ff_af_arls = { |
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.name = "arls", |
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.description = NULL_IF_CONFIG_SMALL("Apply Recursive Least Squares algorithm to first audio stream."), |
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.priv_size = sizeof(AudioRLSContext), |
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.priv_class = &arls_class, |
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.init = init, |
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.uninit = uninit, |
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.activate = activate, |
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FILTER_INPUTS(inputs), |
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FILTER_OUTPUTS(outputs), |
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP), |
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | |
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AVFILTER_FLAG_SLICE_THREADS, |
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.process_command = ff_filter_process_command, |
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}; |
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