|
- /*
- * Copyright (c) 2013 Paul B Mahol
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
- /**
- * @file
- * fade audio filter
- */
-
- #include "libavutil/opt.h"
- #include "audio.h"
- #include "avfilter.h"
- #include "internal.h"
-
- typedef struct {
- const AVClass *class;
- int type;
- int curve;
- int nb_samples;
- int64_t start_sample;
- int64_t duration;
- int64_t start_time;
-
- void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
- int nb_samples, int channels, int direction,
- int64_t start, int range, int curve);
- } AudioFadeContext;
-
- enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, PAR, QUA, CUB, SQU, CBR };
-
- #define OFFSET(x) offsetof(AudioFadeContext, x)
- #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
-
- static const AVOption afade_options[] = {
- { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
- { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
- { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
- { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
- { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
- { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
- { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
- { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
- { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
- { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
- { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
- { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
- { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
- { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
- { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
- { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
- { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
- { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
- { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
- { "par", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
- { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
- { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
- { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
- { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
- { NULL }
- };
-
- AVFILTER_DEFINE_CLASS(afade);
-
- static av_cold int init(AVFilterContext *ctx)
- {
- AudioFadeContext *s = ctx->priv;
-
- if (INT64_MAX - s->nb_samples < s->start_sample)
- return AVERROR(EINVAL);
-
- return 0;
- }
-
- static int query_formats(AVFilterContext *ctx)
- {
- AVFilterFormats *formats;
- AVFilterChannelLayouts *layouts;
- static const enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
- AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
- AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
- AV_SAMPLE_FMT_NONE
- };
-
- layouts = ff_all_channel_layouts();
- if (!layouts)
- return AVERROR(ENOMEM);
- ff_set_common_channel_layouts(ctx, layouts);
-
- formats = ff_make_format_list(sample_fmts);
- if (!formats)
- return AVERROR(ENOMEM);
- ff_set_common_formats(ctx, formats);
-
- formats = ff_all_samplerates();
- if (!formats)
- return AVERROR(ENOMEM);
- ff_set_common_samplerates(ctx, formats);
-
- return 0;
- }
-
- static double fade_gain(int curve, int64_t index, int range)
- {
- double gain;
-
- gain = FFMAX(0.0, FFMIN(1.0, 1.0 * index / range));
-
- switch (curve) {
- case QSIN:
- gain = sin(gain * M_PI / 2.0);
- break;
- case ESIN:
- gain = 1.0 - cos(M_PI / 4.0 * (pow(2.0*gain - 1, 3) + 1));
- break;
- case HSIN:
- gain = (1.0 - cos(gain * M_PI)) / 2.0;
- break;
- case LOG:
- gain = pow(0.1, (1 - gain) * 5.0);
- break;
- case PAR:
- gain = (1 - (1 - gain) * (1 - gain));
- break;
- case QUA:
- gain *= gain;
- break;
- case CUB:
- gain = gain * gain * gain;
- break;
- case SQU:
- gain = sqrt(gain);
- break;
- case CBR:
- gain = cbrt(gain);
- break;
- }
-
- return gain;
- }
-
- #define FADE_PLANAR(name, type) \
- static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
- int nb_samples, int channels, int dir, \
- int64_t start, int range, int curve) \
- { \
- int i, c; \
- \
- for (i = 0; i < nb_samples; i++) { \
- double gain = fade_gain(curve, start + i * dir, range); \
- for (c = 0; c < channels; c++) { \
- type *d = (type *)dst[c]; \
- const type *s = (type *)src[c]; \
- \
- d[i] = s[i] * gain; \
- } \
- } \
- }
-
- #define FADE(name, type) \
- static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
- int nb_samples, int channels, int dir, \
- int64_t start, int range, int curve) \
- { \
- type *d = (type *)dst[0]; \
- const type *s = (type *)src[0]; \
- int i, c, k = 0; \
- \
- for (i = 0; i < nb_samples; i++) { \
- double gain = fade_gain(curve, start + i * dir, range); \
- for (c = 0; c < channels; c++, k++) \
- d[k] = s[k] * gain; \
- } \
- }
-
- FADE_PLANAR(dbl, double)
- FADE_PLANAR(flt, float)
- FADE_PLANAR(s16, int16_t)
- FADE_PLANAR(s32, int32_t)
-
- FADE(dbl, double)
- FADE(flt, float)
- FADE(s16, int16_t)
- FADE(s32, int32_t)
-
- static int config_input(AVFilterLink *inlink)
- {
- AVFilterContext *ctx = inlink->dst;
- AudioFadeContext *s = ctx->priv;
-
- switch (inlink->format) {
- case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
- case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
- case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
- case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
- case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
- case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
- case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
- case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
- }
-
- if (s->duration)
- s->nb_samples = av_rescale(s->duration, inlink->sample_rate, AV_TIME_BASE);
- if (s->start_time)
- s->start_sample = av_rescale(s->start_time, inlink->sample_rate, AV_TIME_BASE);
-
- return 0;
- }
-
- static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
- {
- AudioFadeContext *s = inlink->dst->priv;
- AVFilterLink *outlink = inlink->dst->outputs[0];
- int nb_samples = buf->nb_samples;
- AVFrame *out_buf;
- int64_t cur_sample = av_rescale_q(buf->pts, (AVRational){1, outlink->sample_rate}, outlink->time_base);
-
- if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
- ( s->type && (cur_sample + s->nb_samples < s->start_sample)))
- return ff_filter_frame(outlink, buf);
-
- if (av_frame_is_writable(buf)) {
- out_buf = buf;
- } else {
- out_buf = ff_get_audio_buffer(inlink, nb_samples);
- if (!out_buf)
- return AVERROR(ENOMEM);
- av_frame_copy_props(out_buf, buf);
- }
-
- if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
- ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
- av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
- av_frame_get_channels(out_buf), out_buf->format);
- } else {
- int64_t start;
-
- if (!s->type)
- start = cur_sample - s->start_sample;
- else
- start = s->start_sample + s->nb_samples - cur_sample;
-
- s->fade_samples(out_buf->extended_data, buf->extended_data,
- nb_samples, av_frame_get_channels(buf),
- s->type ? -1 : 1, start,
- s->nb_samples, s->curve);
- }
-
- if (buf != out_buf)
- av_frame_free(&buf);
-
- return ff_filter_frame(outlink, out_buf);
- }
-
- static const AVFilterPad avfilter_af_afade_inputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = filter_frame,
- .config_props = config_input,
- },
- { NULL }
- };
-
- static const AVFilterPad avfilter_af_afade_outputs[] = {
- {
- .name = "default",
- .type = AVMEDIA_TYPE_AUDIO,
- },
- { NULL }
- };
-
- AVFilter ff_af_afade = {
- .name = "afade",
- .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
- .query_formats = query_formats,
- .priv_size = sizeof(AudioFadeContext),
- .init = init,
- .inputs = avfilter_af_afade_inputs,
- .outputs = avfilter_af_afade_outputs,
- .priv_class = &afade_class,
- .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
- };
|