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Diffstat (limited to 'libavfilter/af_afade.c')
| -rw-r--r-- | libavfilter/af_afade.c | 300 |
1 files changed, 300 insertions, 0 deletions
diff --git a/libavfilter/af_afade.c b/libavfilter/af_afade.c new file mode 100644 index 0000000000..806f6f6145 --- /dev/null +++ b/libavfilter/af_afade.c @@ -0,0 +1,300 @@ +/* + * 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, inlink->time_base, (AVRational){1, inlink->sample_rate}); + + 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, +}; |