mirror of
https://github.com/xenia-project/FFmpeg.git
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301 lines
13 KiB
C
301 lines
13 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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/**
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* @file
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* fade audio filter
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*/
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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const AVClass *class;
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int type;
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int curve;
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int nb_samples;
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int64_t start_sample;
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int64_t duration;
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int64_t start_time;
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void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
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int nb_samples, int channels, int direction,
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int64_t start, int range, int curve);
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} AudioFadeContext;
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enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };
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#define OFFSET(x) offsetof(AudioFadeContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption afade_options[] = {
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{ "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
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{ "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
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{ "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
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{ "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
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{ "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
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{ "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
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{ "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
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{ "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
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{ "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
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{ "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
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{ "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
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{ "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
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{ "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
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{ "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
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{ "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
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{ "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
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{ "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
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{ "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
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{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
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{ "par", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
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{ "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
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{ "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
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{ "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
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{ "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(afade);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioFadeContext *s = ctx->priv;
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if (INT64_MAX - s->nb_samples < s->start_sample)
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return AVERROR(EINVAL);
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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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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_set_common_channel_layouts(ctx, layouts);
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_formats(ctx, formats);
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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static double fade_gain(int curve, int64_t index, int range)
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{
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double gain;
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gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
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switch (curve) {
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case QSIN:
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gain = sin(gain * M_PI / 2.0);
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break;
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case ESIN:
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gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
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break;
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case HSIN:
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gain = (1.0 - cos(gain * M_PI)) / 2.0;
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break;
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case LOG:
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gain = pow(0.1, (1 - gain) * 5.0);
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break;
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case PAR:
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gain = (1 - (1 - gain) * (1 - gain));
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break;
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case QUA:
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gain *= gain;
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break;
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case CUB:
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gain = gain * gain * gain;
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break;
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case SQU:
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gain = sqrt(gain);
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break;
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case CBR:
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gain = cbrt(gain);
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break;
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}
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return gain;
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}
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#define FADE_PLANAR(name, type) \
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static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
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int nb_samples, int channels, int dir, \
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int64_t start, int range, int curve) \
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{ \
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int i, c; \
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\
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for (i = 0; i < nb_samples; i++) { \
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double gain = fade_gain(curve, start + i * dir, range); \
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for (c = 0; c < channels; c++) { \
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type *d = (type *)dst[c]; \
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const type *s = (type *)src[c]; \
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\
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d[i] = s[i] * gain; \
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} \
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} \
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}
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#define FADE(name, type) \
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static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
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int nb_samples, int channels, int dir, \
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int64_t start, int range, int curve) \
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{ \
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type *d = (type *)dst[0]; \
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const type *s = (type *)src[0]; \
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int i, c, k = 0; \
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\
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for (i = 0; i < nb_samples; i++) { \
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double gain = fade_gain(curve, start + i * dir, range); \
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for (c = 0; c < channels; c++, k++) \
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d[k] = s[k] * gain; \
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} \
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}
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FADE_PLANAR(dbl, double)
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FADE_PLANAR(flt, float)
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FADE_PLANAR(s16, int16_t)
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FADE_PLANAR(s32, int32_t)
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FADE(dbl, double)
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FADE(flt, float)
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FADE(s16, int16_t)
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FADE(s32, int32_t)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioFadeContext *s = ctx->priv;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
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case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
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case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
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case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
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case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
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case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
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case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
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case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
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}
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if (s->duration)
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s->nb_samples = av_rescale(s->duration, inlink->sample_rate, AV_TIME_BASE);
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if (s->start_time)
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s->start_sample = av_rescale(s->start_time, inlink->sample_rate, AV_TIME_BASE);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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{
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AudioFadeContext *s = inlink->dst->priv;
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AVFilterLink *outlink = inlink->dst->outputs[0];
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int nb_samples = buf->nb_samples;
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AVFrame *out_buf;
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int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
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if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
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( s->type && (cur_sample + s->nb_samples < s->start_sample)))
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return ff_filter_frame(outlink, buf);
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if (av_frame_is_writable(buf)) {
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out_buf = buf;
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} else {
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out_buf = ff_get_audio_buffer(inlink, nb_samples);
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if (!out_buf)
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return AVERROR(ENOMEM);
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av_frame_copy_props(out_buf, buf);
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}
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if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
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( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
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av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
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av_frame_get_channels(out_buf), out_buf->format);
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} else {
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int64_t start;
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if (!s->type)
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start = cur_sample - s->start_sample;
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else
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start = s->start_sample + s->nb_samples - cur_sample;
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s->fade_samples(out_buf->extended_data, buf->extended_data,
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nb_samples, av_frame_get_channels(buf),
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s->type ? -1 : 1, start,
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s->nb_samples, s->curve);
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}
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if (buf != out_buf)
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av_frame_free(&buf);
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return ff_filter_frame(outlink, out_buf);
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}
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static const AVFilterPad avfilter_af_afade_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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};
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static const AVFilterPad avfilter_af_afade_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_afade = {
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.name = "afade",
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.description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(AudioFadeContext),
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.init = init,
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.inputs = avfilter_af_afade_inputs,
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.outputs = avfilter_af_afade_outputs,
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.priv_class = &afade_class,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
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};
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