303 lines
9.6 KiB
C
303 lines
9.6 KiB
C
/*
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* Copyright (c) 2020 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 License
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* 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
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* GNU 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 License
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* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/opt.h"
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#include "libavutil/tx.h"
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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 "internal.h"
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#include "window_func.h"
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typedef struct AudioFIRSourceContext {
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const AVClass *class;
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char *freq_points_str;
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char *magnitude_str;
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char *phase_str;
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int nb_taps;
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int sample_rate;
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int nb_samples;
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int win_func;
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AVComplexFloat *complexf;
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float *freq;
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float *magnitude;
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float *phase;
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int freq_size;
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int magnitude_size;
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int phase_size;
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int nb_freq;
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int nb_magnitude;
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int nb_phase;
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float *taps;
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float *win;
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int64_t pts;
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AVTXContext *tx_ctx;
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av_tx_fn tx_fn;
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} AudioFIRSourceContext;
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#define OFFSET(x) offsetof(AudioFIRSourceContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption afirsrc_options[] = {
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{ "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
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{ "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
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{ "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
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{ "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
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{ "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
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{ "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
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{ "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
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{ "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
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{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
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{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
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{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
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{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
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WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
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WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
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{NULL}
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};
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AVFILTER_DEFINE_CLASS(afirsrc);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioFIRSourceContext *s = ctx->priv;
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if (!(s->nb_taps & 1)) {
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av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);
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s->nb_taps |= 1;
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}
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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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AudioFIRSourceContext *s = ctx->priv;
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av_freep(&s->win);
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av_freep(&s->taps);
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av_freep(&s->freq);
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av_freep(&s->magnitude);
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av_freep(&s->phase);
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av_freep(&s->complexf);
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av_tx_uninit(&s->tx_ctx);
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}
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static av_cold int query_formats(AVFilterContext *ctx)
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{
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AudioFIRSourceContext *s = ctx->priv;
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static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
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int sample_rates[] = { s->sample_rate, -1 };
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE
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};
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int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
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if (ret < 0)
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return ret;
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ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
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if (ret < 0)
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return ret;
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return ff_set_common_samplerates_from_list(ctx, sample_rates);
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}
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static int parse_string(char *str, float **items, int *nb_items, int *items_size)
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{
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float *new_items;
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char *tail;
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new_items = av_fast_realloc(NULL, items_size, 1 * sizeof(float));
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if (!new_items)
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return AVERROR(ENOMEM);
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*items = new_items;
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tail = str;
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if (!tail)
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return AVERROR(EINVAL);
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do {
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(*items)[(*nb_items)++] = av_strtod(tail, &tail);
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new_items = av_fast_realloc(*items, items_size, (*nb_items + 1) * sizeof(float));
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if (!new_items)
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return AVERROR(ENOMEM);
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*items = new_items;
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if (tail && *tail)
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tail++;
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} while (tail && *tail);
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return 0;
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}
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static void lininterp(AVComplexFloat *complexf,
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const float *freq,
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const float *magnitude,
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const float *phase,
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int m, int minterp)
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{
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for (int i = 0; i < minterp; i++) {
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for (int j = 1; j < m; j++) {
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const float x = i / (float)minterp;
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if (x <= freq[j]) {
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const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
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const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
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complexf[i].re = mg * cosf(ph);
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complexf[i].im = mg * sinf(ph);
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break;
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}
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}
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}
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}
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static av_cold int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioFIRSourceContext *s = ctx->priv;
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float overlap, scale = 1.f, compensation;
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int fft_size, middle, ret;
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s->nb_freq = s->nb_magnitude = s->nb_phase = 0;
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ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
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if (ret < 0)
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return ret;
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ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
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if (ret < 0)
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return ret;
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ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);
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if (ret < 0)
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return ret;
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if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {
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av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");
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return AVERROR(EINVAL);
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}
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for (int i = 0; i < s->nb_freq; i++) {
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if (i == 0 && s->freq[i] != 0.f) {
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av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");
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return AVERROR(EINVAL);
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}
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if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {
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av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");
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return AVERROR(EINVAL);
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}
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if (i && s->freq[i] < s->freq[i-1]) {
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av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");
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return AVERROR(EINVAL);
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}
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}
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fft_size = 1 << (av_log2(s->nb_taps) + 1);
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s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));
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if (!s->complexf)
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return AVERROR(ENOMEM);
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ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
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if (ret < 0)
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return ret;
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s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
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if (!s->taps)
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return AVERROR(ENOMEM);
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s->win = av_calloc(s->nb_taps, sizeof(*s->win));
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if (!s->win)
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return AVERROR(ENOMEM);
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generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);
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lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);
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s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(*s->complexf));
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compensation = 2.f / fft_size;
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middle = s->nb_taps / 2;
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for (int i = 0; i <= middle; i++) {
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s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];
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s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i];
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}
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s->pts = 0;
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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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AVFilterLink *outlink = ctx->outputs[0];
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AudioFIRSourceContext *s = ctx->priv;
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AVFrame *frame;
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int nb_samples;
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if (!ff_outlink_frame_wanted(outlink))
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return FFERROR_NOT_READY;
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nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
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if (nb_samples <= 0) {
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ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
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return 0;
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}
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if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
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return AVERROR(ENOMEM);
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memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
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frame->pts = s->pts;
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s->pts += nb_samples;
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return ff_filter_frame(outlink, frame);
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}
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static const AVFilterPad afirsrc_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_asrc_afirsrc = {
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.name = "afirsrc",
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.description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),
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.init = init,
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.uninit = uninit,
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.activate = activate,
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.priv_size = sizeof(AudioFIRSourceContext),
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.inputs = NULL,
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FILTER_OUTPUTS(afirsrc_outputs),
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FILTER_QUERY_FUNC(query_formats),
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.priv_class = &afirsrc_class,
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};
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