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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
353 lines
12 KiB
C
353 lines
12 KiB
C
/*
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* Copyright (c) 2013 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/avassert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "internal.h"
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typedef struct AudioEchoContext {
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const AVClass *class;
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float in_gain, out_gain;
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char *delays, *decays;
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float *delay, *decay;
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int nb_echoes;
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int delay_index;
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uint8_t **delayptrs;
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int max_samples, fade_out;
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int *samples;
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int eof;
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int64_t next_pts;
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void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
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uint8_t * const *src, uint8_t **dst,
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int nb_samples, int channels);
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} AudioEchoContext;
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#define OFFSET(x) offsetof(AudioEchoContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aecho_options[] = {
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{ "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
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{ "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
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{ "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
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{ "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aecho);
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static void count_items(char *item_str, int *nb_items)
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{
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char *p;
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*nb_items = 1;
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for (p = item_str; *p; p++) {
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if (*p == '|')
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(*nb_items)++;
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}
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}
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static void fill_items(char *item_str, int *nb_items, float *items)
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{
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char *p, *saveptr = NULL;
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int i, new_nb_items = 0;
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p = item_str;
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for (i = 0; i < *nb_items; i++) {
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char *tstr = av_strtok(p, "|", &saveptr);
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p = NULL;
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if (tstr)
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new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 1;
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}
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*nb_items = new_nb_items;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioEchoContext *s = ctx->priv;
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av_freep(&s->delay);
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av_freep(&s->decay);
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av_freep(&s->samples);
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if (s->delayptrs)
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av_freep(&s->delayptrs[0]);
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av_freep(&s->delayptrs);
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioEchoContext *s = ctx->priv;
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int nb_delays, nb_decays, i;
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if (!s->delays || !s->decays) {
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av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
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return AVERROR(EINVAL);
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}
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count_items(s->delays, &nb_delays);
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count_items(s->decays, &nb_decays);
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s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
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s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
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if (!s->delay || !s->decay)
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return AVERROR(ENOMEM);
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fill_items(s->delays, &nb_delays, s->delay);
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fill_items(s->decays, &nb_decays, s->decay);
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if (nb_delays != nb_decays) {
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av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
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return AVERROR(EINVAL);
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}
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s->nb_echoes = nb_delays;
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if (!s->nb_echoes) {
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av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
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return AVERROR(EINVAL);
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}
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s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
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if (!s->samples)
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return AVERROR(ENOMEM);
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for (i = 0; i < nb_delays; i++) {
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if (s->delay[i] <= 0 || s->delay[i] > 90000) {
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av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
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return AVERROR(EINVAL);
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}
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if (s->decay[i] <= 0 || s->decay[i] > 1) {
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av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
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return AVERROR(EINVAL);
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}
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}
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s->next_pts = AV_NOPTS_VALUE;
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av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
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return 0;
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}
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#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
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#define ECHO(name, type, min, max) \
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static void echo_samples_## name ##p(AudioEchoContext *ctx, \
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uint8_t **delayptrs, \
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uint8_t * const *src, uint8_t **dst, \
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int nb_samples, int channels) \
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{ \
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const double out_gain = ctx->out_gain; \
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const double in_gain = ctx->in_gain; \
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const int nb_echoes = ctx->nb_echoes; \
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const int max_samples = ctx->max_samples; \
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int i, j, chan, av_uninit(index); \
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\
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av_assert1(channels > 0); /* would corrupt delay_index */ \
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\
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for (chan = 0; chan < channels; chan++) { \
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const type *s = (type *)src[chan]; \
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type *d = (type *)dst[chan]; \
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type *dbuf = (type *)delayptrs[chan]; \
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\
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index = ctx->delay_index; \
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for (i = 0; i < nb_samples; i++, s++, d++) { \
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double out, in; \
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\
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in = *s; \
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out = in * in_gain; \
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for (j = 0; j < nb_echoes; j++) { \
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int ix = index + max_samples - ctx->samples[j]; \
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ix = MOD(ix, max_samples); \
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out += dbuf[ix] * ctx->decay[j]; \
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} \
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out *= out_gain; \
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\
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*d = av_clipd(out, min, max); \
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dbuf[index] = in; \
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\
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index = MOD(index + 1, max_samples); \
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} \
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} \
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ctx->delay_index = index; \
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}
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ECHO(dbl, double, -1.0, 1.0 )
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ECHO(flt, float, -1.0, 1.0 )
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ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
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ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
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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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AudioEchoContext *s = ctx->priv;
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float volume = 1.0;
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int i;
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for (i = 0; i < s->nb_echoes; i++) {
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s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
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s->max_samples = FFMAX(s->max_samples, s->samples[i]);
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volume += s->decay[i];
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}
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if (s->max_samples <= 0) {
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av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
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return AVERROR(EINVAL);
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}
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s->fade_out = s->max_samples;
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if (volume * s->in_gain * s->out_gain > 1.0)
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av_log(ctx, AV_LOG_WARNING,
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"out_gain %f can cause saturation of output\n", s->out_gain);
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switch (outlink->format) {
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case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
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case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
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case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
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case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
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}
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if (s->delayptrs)
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av_freep(&s->delayptrs[0]);
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av_freep(&s->delayptrs);
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return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
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outlink->ch_layout.nb_channels,
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s->max_samples,
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outlink->format, 0);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioEchoContext *s = ctx->priv;
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AVFrame *out_frame;
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if (av_frame_is_writable(frame)) {
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out_frame = frame;
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} else {
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out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
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if (!out_frame) {
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av_frame_free(&frame);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out_frame, frame);
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}
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s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
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frame->nb_samples, inlink->ch_layout.nb_channels);
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s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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if (frame != out_frame)
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av_frame_free(&frame);
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return ff_filter_frame(ctx->outputs[0], out_frame);
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioEchoContext *s = ctx->priv;
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int nb_samples = FFMIN(s->fade_out, 2048);
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AVFrame *frame = ff_get_audio_buffer(outlink, nb_samples);
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if (!frame)
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return AVERROR(ENOMEM);
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s->fade_out -= nb_samples;
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av_samples_set_silence(frame->extended_data, 0,
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frame->nb_samples,
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outlink->ch_layout.nb_channels,
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frame->format);
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s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
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frame->nb_samples, outlink->ch_layout.nb_channels);
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frame->pts = s->next_pts;
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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return ff_filter_frame(outlink, frame);
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}
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static int activate(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AudioEchoContext *s = ctx->priv;
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AVFrame *in;
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int ret, status;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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ret = ff_inlink_consume_frame(inlink, &in);
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if (ret < 0)
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return ret;
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if (ret > 0)
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return filter_frame(inlink, in);
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if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
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if (status == AVERROR_EOF)
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s->eof = 1;
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}
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if (s->eof && s->fade_out <= 0) {
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ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
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return 0;
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}
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if (!s->eof)
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FF_FILTER_FORWARD_WANTED(outlink, inlink);
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return request_frame(outlink);
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}
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static const AVFilterPad aecho_outputs[] = {
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{
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.name = "default",
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.config_props = config_output,
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_af_aecho = {
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.name = "aecho",
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.description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
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.priv_size = sizeof(AudioEchoContext),
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.priv_class = &aecho_class,
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.init = init,
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.activate = activate,
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.uninit = uninit,
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FILTER_INPUTS(ff_audio_default_filterpad),
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FILTER_OUTPUTS(aecho_outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
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};
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