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/* |
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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/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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✓✓ |
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for (p = item_str; *p; p++) { |
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✗✓ |
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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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✓✓ |
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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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✓✗ |
2 |
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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✓✗ |
1 |
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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✓✗✗✓
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1 |
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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✓✗✗✓
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1 |
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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✗✓ |
1 |
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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✗✓ |
1 |
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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✗✓ |
1 |
if (!s->samples) |
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return AVERROR(ENOMEM); |
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✓✓ |
2 |
for (i = 0; i < nb_delays; i++) { |
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✓✗✗✓
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1 |
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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✓✗✗✓
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1 |
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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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterChannelLayouts *layouts; |
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AVFilterFormats *formats; |
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static const enum AVSampleFormat sample_fmts[] = { |
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AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, |
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, |
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AV_SAMPLE_FMT_NONE |
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}; |
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int ret; |
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layouts = ff_all_channel_counts(); |
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✗✓ |
1 |
if (!layouts) |
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return AVERROR(ENOMEM); |
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ret = ff_set_common_channel_layouts(ctx, layouts); |
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✗✓ |
1 |
if (ret < 0) |
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return ret; |
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formats = ff_make_format_list(sample_fmts); |
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✗✓ |
1 |
if (!formats) |
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return AVERROR(ENOMEM); |
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ret = ff_set_common_formats(ctx, formats); |
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✗✓ |
1 |
if (ret < 0) |
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return ret; |
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formats = ff_all_samplerates(); |
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✗✓ |
1 |
if (!formats) |
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return AVERROR(ENOMEM); |
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return ff_set_common_samplerates(ctx, formats); |
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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; \ |
213 |
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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); \ |
216 |
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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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\ |
220 |
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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 ) |
230 |
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ECHO(flt, float, -1.0, 1.0 ) |
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✗✓✓✓ ✓✓✓✓ ✓✓ |
1064824 |
ECHO(s16, int16_t, INT16_MIN, INT16_MAX) |
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ECHO(s32, int32_t, INT32_MIN, INT32_MAX) |
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1 |
static int config_output(AVFilterLink *outlink) |
235 |
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{ |
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1 |
AVFilterContext *ctx = outlink->src; |
237 |
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AudioEchoContext *s = ctx->priv; |
238 |
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1 |
float volume = 1.0; |
239 |
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int i; |
240 |
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241 |
✓✓ |
2 |
for (i = 0; i < s->nb_echoes; i++) { |
242 |
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1 |
s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0; |
243 |
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1 |
s->max_samples = FFMAX(s->max_samples, s->samples[i]); |
244 |
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1 |
volume += s->decay[i]; |
245 |
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} |
246 |
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247 |
✗✓ |
1 |
if (s->max_samples <= 0) { |
248 |
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av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n"); |
249 |
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return AVERROR(EINVAL); |
250 |
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} |
251 |
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1 |
s->fade_out = s->max_samples; |
252 |
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253 |
✗✓ |
1 |
if (volume * s->in_gain * s->out_gain > 1.0) |
254 |
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av_log(ctx, AV_LOG_WARNING, |
255 |
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"out_gain %f can cause saturation of output\n", s->out_gain); |
256 |
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257 |
✗✗✓✗ ✗ |
1 |
switch (outlink->format) { |
258 |
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case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break; |
259 |
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case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break; |
260 |
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1 |
case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break; |
261 |
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case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break; |
262 |
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} |
263 |
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264 |
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265 |
✗✓ |
1 |
if (s->delayptrs) |
266 |
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av_freep(&s->delayptrs[0]); |
267 |
|
1 |
av_freep(&s->delayptrs); |
268 |
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269 |
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1 |
return av_samples_alloc_array_and_samples(&s->delayptrs, NULL, |
270 |
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outlink->channels, |
271 |
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s->max_samples, |
272 |
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1 |
outlink->format, 0); |
273 |
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} |
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275 |
|
259 |
static int filter_frame(AVFilterLink *inlink, AVFrame *frame) |
276 |
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{ |
277 |
|
259 |
AVFilterContext *ctx = inlink->dst; |
278 |
|
259 |
AudioEchoContext *s = ctx->priv; |
279 |
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AVFrame *out_frame; |
280 |
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|
281 |
✓✗ |
259 |
if (av_frame_is_writable(frame)) { |
282 |
|
259 |
out_frame = frame; |
283 |
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} else { |
284 |
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out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples); |
285 |
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if (!out_frame) { |
286 |
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av_frame_free(&frame); |
287 |
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return AVERROR(ENOMEM); |
288 |
|
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} |
289 |
|
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av_frame_copy_props(out_frame, frame); |
290 |
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} |
291 |
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292 |
|
259 |
s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data, |
293 |
|
259 |
frame->nb_samples, inlink->channels); |
294 |
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295 |
|
259 |
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); |
296 |
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|
297 |
✗✓ |
259 |
if (frame != out_frame) |
298 |
|
|
av_frame_free(&frame); |
299 |
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|
300 |
|
259 |
return ff_filter_frame(ctx->outputs[0], out_frame); |
301 |
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} |
302 |
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|
303 |
|
1 |
static int request_frame(AVFilterLink *outlink) |
304 |
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{ |
305 |
|
1 |
AVFilterContext *ctx = outlink->src; |
306 |
|
1 |
AudioEchoContext *s = ctx->priv; |
307 |
|
1 |
int nb_samples = FFMIN(s->fade_out, 2048); |
308 |
|
1 |
AVFrame *frame = ff_get_audio_buffer(outlink, nb_samples); |
309 |
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|
310 |
✗✓ |
1 |
if (!frame) |
311 |
|
|
return AVERROR(ENOMEM); |
312 |
|
1 |
s->fade_out -= nb_samples; |
313 |
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|
314 |
|
1 |
av_samples_set_silence(frame->extended_data, 0, |
315 |
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frame->nb_samples, |
316 |
|
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outlink->channels, |
317 |
|
1 |
frame->format); |
318 |
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|
319 |
|
1 |
s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data, |
320 |
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frame->nb_samples, outlink->channels); |
321 |
|
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|
322 |
|
1 |
frame->pts = s->next_pts; |
323 |
✓✗ |
1 |
if (s->next_pts != AV_NOPTS_VALUE) |
324 |
|
1 |
s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
325 |
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|
326 |
|
1 |
return ff_filter_frame(outlink, frame); |
327 |
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} |
328 |
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|
329 |
|
521 |
static int activate(AVFilterContext *ctx) |
330 |
|
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{ |
331 |
|
521 |
AVFilterLink *inlink = ctx->inputs[0]; |
332 |
|
521 |
AVFilterLink *outlink = ctx->outputs[0]; |
333 |
|
521 |
AudioEchoContext *s = ctx->priv; |
334 |
|
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AVFrame *in; |
335 |
|
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int ret, status; |
336 |
|
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int64_t pts; |
337 |
|
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|
338 |
✓✓ |
521 |
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); |
339 |
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|
340 |
|
520 |
ret = ff_inlink_consume_frame(inlink, &in); |
341 |
✗✓ |
520 |
if (ret < 0) |
342 |
|
|
return ret; |
343 |
✓✓ |
520 |
if (ret > 0) |
344 |
|
259 |
return filter_frame(inlink, in); |
345 |
|
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|
346 |
✓✓✓✓
|
261 |
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { |
347 |
✓✗ |
1 |
if (status == AVERROR_EOF) |
348 |
|
1 |
s->eof = 1; |
349 |
|
|
} |
350 |
|
|
|
351 |
✓✓✓✓
|
261 |
if (s->eof && s->fade_out <= 0) { |
352 |
|
1 |
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts); |
353 |
|
1 |
return 0; |
354 |
|
|
} |
355 |
|
|
|
356 |
✓✓ |
260 |
if (!s->eof) |
357 |
✓✗ |
259 |
FF_FILTER_FORWARD_WANTED(outlink, inlink); |
358 |
|
|
|
359 |
|
1 |
return request_frame(outlink); |
360 |
|
|
} |
361 |
|
|
|
362 |
|
|
static const AVFilterPad aecho_inputs[] = { |
363 |
|
|
{ |
364 |
|
|
.name = "default", |
365 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
366 |
|
|
}, |
367 |
|
|
{ NULL } |
368 |
|
|
}; |
369 |
|
|
|
370 |
|
|
static const AVFilterPad aecho_outputs[] = { |
371 |
|
|
{ |
372 |
|
|
.name = "default", |
373 |
|
|
.config_props = config_output, |
374 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
375 |
|
|
}, |
376 |
|
|
{ NULL } |
377 |
|
|
}; |
378 |
|
|
|
379 |
|
|
AVFilter ff_af_aecho = { |
380 |
|
|
.name = "aecho", |
381 |
|
|
.description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."), |
382 |
|
|
.query_formats = query_formats, |
383 |
|
|
.priv_size = sizeof(AudioEchoContext), |
384 |
|
|
.priv_class = &aecho_class, |
385 |
|
|
.init = init, |
386 |
|
|
.activate = activate, |
387 |
|
|
.uninit = uninit, |
388 |
|
|
.inputs = aecho_inputs, |
389 |
|
|
.outputs = aecho_outputs, |
390 |
|
|
}; |