/* * Copyright (c) 2019 Paul B Mahol * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/float_dsp.h" #include "libavutil/opt.h" #include "audio.h" #include "avfilter.h" #include "formats.h" #include "filters.h" #include "internal.h" enum OutModes { IN_MODE, DESIRED_MODE, OUT_MODE, NOISE_MODE, NB_OMODES }; typedef struct AudioNLMSContext { const AVClass *class; int order; float mu; float eps; float leakage; int output_mode; int kernel_size; AVFrame *offset; AVFrame *delay; AVFrame *coeffs; AVFrame *tmp; AVFrame *frame[2]; AVFloatDSPContext *fdsp; } AudioNLMSContext; #define OFFSET(x) offsetof(AudioNLMSContext, x) #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM #define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption anlms_options[] = { { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A }, { "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 2, AT }, { "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AT }, { "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, AT }, { "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" }, { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" }, { "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" }, { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" }, { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" }, { NULL } }; AVFILTER_DEFINE_CLASS(anlms); static int query_formats(AVFilterContext *ctx) { static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }; int ret = ff_set_common_all_channel_counts(ctx); if (ret < 0) return ret; ret = ff_set_common_formats_from_list(ctx, sample_fmts); if (ret < 0) return ret; return ff_set_common_all_samplerates(ctx); } static float fir_sample(AudioNLMSContext *s, float sample, float *delay, float *coeffs, float *tmp, int *offset) { const int order = s->order; float output; delay[*offset] = sample; memcpy(tmp, coeffs + order - *offset, order * sizeof(float)); output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size); if (--(*offset) < 0) *offset = order - 1; return output; } static float process_sample(AudioNLMSContext *s, float input, float desired, float *delay, float *coeffs, float *tmp, int *offsetp) { const int order = s->order; const float leakage = s->leakage; const float mu = s->mu; const float a = 1.f - leakage * mu; float sum, output, e, norm, b; int offset = *offsetp; delay[offset + order] = input; output = fir_sample(s, input, delay, coeffs, tmp, offsetp); e = desired - output; sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size); norm = s->eps + sum; b = mu * e / norm; memcpy(tmp, delay + offset, order * sizeof(float)); s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size); s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size); memcpy(coeffs + order, coeffs, order * sizeof(float)); switch (s->output_mode) { case IN_MODE: output = input; break; case DESIRED_MODE: output = desired; break; case OUT_MODE: /*output = output;*/ break; case NOISE_MODE: output = desired - output; break; } return output; } static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AudioNLMSContext *s = ctx->priv; AVFrame *out = arg; const int start = (out->channels * jobnr) / nb_jobs; const int end = (out->channels * (jobnr+1)) / nb_jobs; for (int c = start; c < end; c++) { const float *input = (const float *)s->frame[0]->extended_data[c]; const float *desired = (const float *)s->frame[1]->extended_data[c]; float *delay = (float *)s->delay->extended_data[c]; float *coeffs = (float *)s->coeffs->extended_data[c]; float *tmp = (float *)s->tmp->extended_data[c]; int *offset = (int *)s->offset->extended_data[c]; float *output = (float *)out->extended_data[c]; for (int n = 0; n < out->nb_samples; n++) output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset); } return 0; } static int activate(AVFilterContext *ctx) { AudioNLMSContext *s = ctx->priv; int i, ret, status; int nb_samples; int64_t pts; FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1])); for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) { if (s->frame[i]) continue; if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) { ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]); if (ret < 0) return ret; } } if (s->frame[0] && s->frame[1]) { AVFrame *out; out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples); if (!out) { av_frame_free(&s->frame[0]); av_frame_free(&s->frame[1]); return AVERROR(ENOMEM); } ff_filter_execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels, ff_filter_get_nb_threads(ctx))); out->pts = s->frame[0]->pts; av_frame_free(&s->frame[0]); av_frame_free(&s->frame[1]); ret = ff_filter_frame(ctx->outputs[0], out); if (ret < 0) return ret; } if (!nb_samples) { for (i = 0; i < 2; i++) { if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { ff_outlink_set_status(ctx->outputs[0], status, pts); return 0; } } } if (ff_outlink_frame_wanted(ctx->outputs[0])) { for (i = 0; i < 2; i++) { if (ff_inlink_queued_samples(ctx->inputs[i]) > 0) continue; ff_inlink_request_frame(ctx->inputs[i]); return 0; } } return 0; } static int config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AudioNLMSContext *s = ctx->priv; s->kernel_size = FFALIGN(s->order, 16); if (!s->offset) s->offset = ff_get_audio_buffer(outlink, 1); if (!s->delay) s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size); if (!s->coeffs) s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size); if (!s->tmp) s->tmp = ff_get_audio_buffer(outlink, s->kernel_size); if (!s->delay || !s->coeffs || !s->offset || !s->tmp) return AVERROR(ENOMEM); return 0; } static av_cold int init(AVFilterContext *ctx) { AudioNLMSContext *s = ctx->priv; s->fdsp = avpriv_float_dsp_alloc(0); if (!s->fdsp) return AVERROR(ENOMEM); return 0; } static av_cold void uninit(AVFilterContext *ctx) { AudioNLMSContext *s = ctx->priv; av_freep(&s->fdsp); av_frame_free(&s->delay); av_frame_free(&s->coeffs); av_frame_free(&s->offset); av_frame_free(&s->tmp); } static const AVFilterPad inputs[] = { { .name = "input", .type = AVMEDIA_TYPE_AUDIO, }, { .name = "desired", .type = AVMEDIA_TYPE_AUDIO, }, { NULL } }; static const AVFilterPad outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }, { NULL } }; const AVFilter ff_af_anlms = { .name = "anlms", .description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."), .priv_size = sizeof(AudioNLMSContext), .priv_class = &anlms_class, .init = init, .uninit = uninit, .activate = activate, .query_formats = query_formats, .inputs = inputs, .outputs = outputs, .flags = AVFILTER_FLAG_SLICE_THREADS, .process_command = ff_filter_process_command, };