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avfilter/af_anlms: add double sample format support
This commit is contained in:
Paul B Mahol
parent
ea6817d2a7
commit
42e45ea8ff
@ -2687,6 +2687,20 @@ Pass error signal estimated samples.
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Default value is @var{o}.
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@end table
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@item precision
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Set which precision to use when processing samples.
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@table @option
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@item auto
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Auto pick internal sample format depending on other filters.
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@item float
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Always use single-floating point precision sample format.
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@item double
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Always use double-floating point precision sample format.
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@end table
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@end table
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@subsection Examples
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@ -26,6 +26,7 @@
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "formats.h"
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#include "internal.h"
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enum OutModes {
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@ -45,6 +46,7 @@ typedef struct AudioNLMSContext {
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float eps;
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float leakage;
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int output_mode;
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int precision;
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int kernel_size;
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AVFrame *offset;
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@ -56,6 +58,8 @@ typedef struct AudioNLMSContext {
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int anlmf;
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int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
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AVFloatDSPContext *fdsp;
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} AudioNLMSContext;
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@ -74,93 +78,32 @@ static const AVOption anlms_options[] = {
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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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{ "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, "mode" },
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{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "precision" },
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{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "precision" },
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{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "precision" },
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{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "precision" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS_EXT(anlms, "anlm(f|s)", anlms_options);
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static float fir_sample(AudioNLMSContext *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(AudioNLMSContext *s, float input, float desired,
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float *delay, float *coeffs, float *tmp, int *offsetp)
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{
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const int order = s->order;
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const float leakage = s->leakage;
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const float mu = s->mu;
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const float a = 1.f - leakage;
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float sum, output, e, norm, b;
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int offset = *offsetp;
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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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sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
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norm = s->eps + sum;
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b = mu * e / norm;
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if (s->anlmf)
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b *= e * e;
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memcpy(tmp, delay + offset, order * sizeof(float));
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s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
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s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
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memcpy(coeffs + order, coeffs, order * sizeof(float));
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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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case ERROR_MODE: 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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static int query_formats(AVFilterContext *ctx)
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{
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AudioNLMSContext *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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static const enum AVSampleFormat sample_fmts[3][3] = {
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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};
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int ret;
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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 *delay = (float *)s->delay->extended_data[c];
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float *coeffs = (float *)s->coeffs->extended_data[c];
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float *tmp = (float *)s->tmp->extended_data[c];
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int *offset = (int *)s->offset->extended_data[c];
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float *output = (float *)out->extended_data[c];
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if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
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return ret;
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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], delay, coeffs, tmp, offset);
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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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if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
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return ret;
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return 0;
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return ff_set_common_all_samplerates(ctx);
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}
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static int activate(AVFilterContext *ctx)
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@ -195,7 +138,7 @@ static int activate(AVFilterContext *ctx)
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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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ff_filter_execute(ctx, s->filter_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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@ -228,6 +171,13 @@ static int activate(AVFilterContext *ctx)
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return 0;
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}
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#define DEPTH 32
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#include "anlms_template.c"
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#undef DEPTH
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#define DEPTH 64
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#include "anlms_template.c"
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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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@ -247,6 +197,15 @@ static int config_output(AVFilterLink *outlink)
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if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
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return AVERROR(ENOMEM);
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switch (outlink->format) {
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case AV_SAMPLE_FMT_DBLP:
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s->filter_channels = filter_channels_double;
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break;
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case AV_SAMPLE_FMT_FLTP:
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s->filter_channels = filter_channels_float;
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break;
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}
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return 0;
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}
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@ -317,7 +276,7 @@ const AVFilter ff_af_anlmf = {
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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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FILTER_QUERY_FUNC(query_formats),
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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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141
libavfilter/anlms_template.c
Normal file
141
libavfilter/anlms_template.c
Normal file
@ -0,0 +1,141 @@
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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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#undef ONE
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#undef ftype
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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 ftype float
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#define ONE 1.f
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#else
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#define SAMPLE_FORMAT double
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#define ftype double
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#define ONE 1.0
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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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#if DEPTH == 64
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static double scalarproduct_double(const double *v1, const double *v2, int len)
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{
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double p = 0.0;
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for (int i = 0; i < len; i++)
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p += v1[i] * v2[i];
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return p;
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}
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#endif
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static ftype fn(fir_sample)(AudioNLMSContext *s, ftype sample, ftype *delay,
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ftype *coeffs, ftype *tmp, int *offset)
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{
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const int order = s->order;
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ftype output;
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delay[*offset] = sample;
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memcpy(tmp, coeffs + order - *offset, order * sizeof(ftype));
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#if DEPTH == 32
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output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
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#else
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output = scalarproduct_double(delay, tmp, s->kernel_size);
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#endif
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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 ftype fn(process_sample)(AudioNLMSContext *s, ftype input, ftype desired,
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ftype *delay, ftype *coeffs, ftype *tmp, int *offsetp)
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{
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const int order = s->order;
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const ftype leakage = s->leakage;
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const ftype mu = s->mu;
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const ftype a = ONE - leakage;
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ftype sum, output, e, norm, b;
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int offset = *offsetp;
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delay[offset + order] = input;
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output = fn(fir_sample)(s, input, delay, coeffs, tmp, offsetp);
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e = desired - output;
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#if DEPTH == 32
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sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
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#else
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sum = scalarproduct_double(delay, delay, s->kernel_size);
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#endif
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norm = s->eps + sum;
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b = mu * e / norm;
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if (s->anlmf)
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b *= e * e;
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memcpy(tmp, delay + offset, order * sizeof(ftype));
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
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s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
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#else
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s->fdsp->vector_dmul_scalar(coeffs, coeffs, a, s->kernel_size);
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s->fdsp->vector_dmac_scalar(coeffs, tmp, b, s->kernel_size);
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#endif
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memcpy(coeffs + order, coeffs, order * sizeof(ftype));
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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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case ERROR_MODE: break;
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}
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return output;
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}
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static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioNLMSContext *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 ftype *input = (const ftype *)s->frame[0]->extended_data[c];
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const ftype *desired = (const ftype *)s->frame[1]->extended_data[c];
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ftype *delay = (ftype *)s->delay->extended_data[c];
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ftype *coeffs = (ftype *)s->coeffs->extended_data[c];
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ftype *tmp = (ftype *)s->tmp->extended_data[c];
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int *offset = (int *)s->offset->extended_data[c];
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ftype *output = (ftype *)out->extended_data[c];
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for (int n = 0; n < out->nb_samples; n++) {
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output[n] = fn(process_sample)(s, input[n], desired[n], delay, coeffs, tmp, offset);
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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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