From 42e45ea8ff30608fb4a86f247a2e4553ff6bf8fe Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Mon, 27 Nov 2023 19:53:38 +0100 Subject: [PATCH] avfilter/af_anlms: add double sample format support --- doc/filters.texi | 14 ++++ libavfilter/af_anlms.c | 117 ++++++++++------------------- libavfilter/anlms_template.c | 141 +++++++++++++++++++++++++++++++++++ 3 files changed, 193 insertions(+), 79 deletions(-) create mode 100644 libavfilter/anlms_template.c diff --git a/doc/filters.texi b/doc/filters.texi index 80ffbb2c65..83c48fe367 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2687,6 +2687,20 @@ Pass error signal estimated samples. Default value is @var{o}. @end table + +@item precision +Set which precision to use when processing samples. + +@table @option +@item auto +Auto pick internal sample format depending on other filters. + +@item float +Always use single-floating point precision sample format. + +@item double +Always use double-floating point precision sample format. +@end table @end table @subsection Examples diff --git a/libavfilter/af_anlms.c b/libavfilter/af_anlms.c index 3191ed1b31..9d3c44575b 100644 --- a/libavfilter/af_anlms.c +++ b/libavfilter/af_anlms.c @@ -26,6 +26,7 @@ #include "audio.h" #include "avfilter.h" #include "filters.h" +#include "formats.h" #include "internal.h" enum OutModes { @@ -45,6 +46,7 @@ typedef struct AudioNLMSContext { float eps; float leakage; int output_mode; + int precision; int kernel_size; AVFrame *offset; @@ -56,6 +58,8 @@ typedef struct AudioNLMSContext { int anlmf; + int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); + AVFloatDSPContext *fdsp; } AudioNLMSContext; @@ -74,93 +78,32 @@ static const AVOption anlms_options[] = { { "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" }, { "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, "mode" }, + { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "precision" }, + { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "precision" }, + { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "precision" }, + { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "precision" }, { NULL } }; AVFILTER_DEFINE_CLASS_EXT(anlms, "anlm(f|s)", anlms_options); -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; - 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; - if (s->anlmf) - b *= e * e; - - 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 = desired - output; break; - case NOISE_MODE: output = input - output; break; - case ERROR_MODE: break; - } - return output; -} - -static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) +static int query_formats(AVFilterContext *ctx) { AudioNLMSContext *s = ctx->priv; - AVFrame *out = arg; - const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs; - const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; + static const enum AVSampleFormat sample_fmts[3][3] = { + { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, + { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, + { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, + }; + int ret; - 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]; + if ((ret = ff_set_common_all_channel_counts(ctx)) < 0) + return ret; - for (int n = 0; n < out->nb_samples; n++) { - output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset); - if (ctx->is_disabled) - output[n] = input[n]; - } - } + if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0) + return ret; - return 0; + return ff_set_common_all_samplerates(ctx); } static int activate(AVFilterContext *ctx) @@ -195,7 +138,7 @@ static int activate(AVFilterContext *ctx) return AVERROR(ENOMEM); } - ff_filter_execute(ctx, process_channels, out, NULL, + ff_filter_execute(ctx, s->filter_channels, out, NULL, FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); out->pts = s->frame[0]->pts; @@ -228,6 +171,13 @@ static int activate(AVFilterContext *ctx) return 0; } +#define DEPTH 32 +#include "anlms_template.c" + +#undef DEPTH +#define DEPTH 64 +#include "anlms_template.c" + static int config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; @@ -247,6 +197,15 @@ static int config_output(AVFilterLink *outlink) if (!s->delay || !s->coeffs || !s->offset || !s->tmp) return AVERROR(ENOMEM); + switch (outlink->format) { + case AV_SAMPLE_FMT_DBLP: + s->filter_channels = filter_channels_double; + break; + case AV_SAMPLE_FMT_FLTP: + s->filter_channels = filter_channels_float; + break; + } + return 0; } @@ -317,7 +276,7 @@ const AVFilter ff_af_anlmf = { .activate = activate, FILTER_INPUTS(inputs), FILTER_OUTPUTS(outputs), - FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP), + FILTER_QUERY_FUNC(query_formats), .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | AVFILTER_FLAG_SLICE_THREADS, .process_command = ff_filter_process_command, diff --git a/libavfilter/anlms_template.c b/libavfilter/anlms_template.c new file mode 100644 index 0000000000..b25df4fa18 --- /dev/null +++ b/libavfilter/anlms_template.c @@ -0,0 +1,141 @@ +/* + * 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 + */ + +#undef ONE +#undef ftype +#undef SAMPLE_FORMAT +#if DEPTH == 32 +#define SAMPLE_FORMAT float +#define ftype float +#define ONE 1.f +#else +#define SAMPLE_FORMAT double +#define ftype double +#define ONE 1.0 +#endif + +#define fn3(a,b) a##_##b +#define fn2(a,b) fn3(a,b) +#define fn(a) fn2(a, SAMPLE_FORMAT) + +#if DEPTH == 64 +static double scalarproduct_double(const double *v1, const double *v2, int len) +{ + double p = 0.0; + + for (int i = 0; i < len; i++) + p += v1[i] * v2[i]; + + return p; +} +#endif + +static ftype fn(fir_sample)(AudioNLMSContext *s, ftype sample, ftype *delay, + ftype *coeffs, ftype *tmp, int *offset) +{ + const int order = s->order; + ftype output; + + delay[*offset] = sample; + + memcpy(tmp, coeffs + order - *offset, order * sizeof(ftype)); + +#if DEPTH == 32 + output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size); +#else + output = scalarproduct_double(delay, tmp, s->kernel_size); +#endif + + if (--(*offset) < 0) + *offset = order - 1; + + return output; +} + +static ftype fn(process_sample)(AudioNLMSContext *s, ftype input, ftype desired, + ftype *delay, ftype *coeffs, ftype *tmp, int *offsetp) +{ + const int order = s->order; + const ftype leakage = s->leakage; + const ftype mu = s->mu; + const ftype a = ONE - leakage; + ftype sum, output, e, norm, b; + int offset = *offsetp; + + delay[offset + order] = input; + + output = fn(fir_sample)(s, input, delay, coeffs, tmp, offsetp); + e = desired - output; + +#if DEPTH == 32 + sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size); +#else + sum = scalarproduct_double(delay, delay, s->kernel_size); +#endif + norm = s->eps + sum; + b = mu * e / norm; + if (s->anlmf) + b *= e * e; + + memcpy(tmp, delay + offset, order * sizeof(ftype)); + +#if DEPTH == 32 + s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size); + s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size); +#else + s->fdsp->vector_dmul_scalar(coeffs, coeffs, a, s->kernel_size); + s->fdsp->vector_dmac_scalar(coeffs, tmp, b, s->kernel_size); +#endif + + memcpy(coeffs + order, coeffs, order * sizeof(ftype)); + + switch (s->output_mode) { + case IN_MODE: output = input; break; + case DESIRED_MODE: output = desired; break; + case OUT_MODE: output = desired - output; break; + case NOISE_MODE: output = input - output; break; + case ERROR_MODE: break; + } + return output; +} + +static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) +{ + AudioNLMSContext *s = ctx->priv; + AVFrame *out = arg; + const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs; + const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; + + for (int c = start; c < end; c++) { + const ftype *input = (const ftype *)s->frame[0]->extended_data[c]; + const ftype *desired = (const ftype *)s->frame[1]->extended_data[c]; + ftype *delay = (ftype *)s->delay->extended_data[c]; + ftype *coeffs = (ftype *)s->coeffs->extended_data[c]; + ftype *tmp = (ftype *)s->tmp->extended_data[c]; + int *offset = (int *)s->offset->extended_data[c]; + ftype *output = (ftype *)out->extended_data[c]; + + for (int n = 0; n < out->nb_samples; n++) { + output[n] = fn(process_sample)(s, input[n], desired[n], delay, coeffs, tmp, offset); + if (ctx->is_disabled) + output[n] = input[n]; + } + } + + return 0; +}