/* * Copyright (c) 2013-2015 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 "config_components.h" #include "libavutil/opt.h" #include "audio.h" #include "avfilter.h" #include "filters.h" #include "internal.h" typedef struct AudioFadeContext { const AVClass *class; int type; int curve, curve2; int64_t nb_samples; int64_t start_sample; int64_t duration; int64_t start_time; double silence; double unity; int overlap; int cf0_eof; int crossfade_is_over; int64_t pts; void (*fade_samples)(uint8_t **dst, uint8_t * const *src, int nb_samples, int channels, int direction, int64_t start, int64_t range, int curve, double silence, double unity); void (*scale_samples)(uint8_t **dst, uint8_t * const *src, int nb_samples, int channels, double unity); void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0, uint8_t * const *cf1, int nb_samples, int channels, int curve0, int curve1); } AudioFadeContext; enum CurveType { NONE = -1, TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, LOSI, SINC, ISINC, NB_CURVES }; #define OFFSET(x) offsetof(AudioFadeContext, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM #define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM 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 }; static double fade_gain(int curve, int64_t index, int64_t range, double silence, double unity) { #define CUBE(a) ((a)*(a)*(a)) double gain; gain = av_clipd(1.0 * index / range, 0, 1.0); switch (curve) { case QSIN: gain = sin(gain * M_PI / 2.0); break; case IQSIN: /* 0.6... = 2 / M_PI */ gain = 0.6366197723675814 * asin(gain); break; case ESIN: gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1)); break; case HSIN: gain = (1.0 - cos(gain * M_PI)) / 2.0; break; case IHSIN: /* 0.3... = 1 / M_PI */ gain = 0.3183098861837907 * acos(1 - 2 * gain); break; case EXP: /* -11.5... = 5*ln(0.1) */ gain = exp(-11.512925464970227 * (1 - gain)); break; case LOG: gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0); break; case PAR: gain = 1 - sqrt(1 - gain); break; case IPAR: gain = (1 - (1 - gain) * (1 - gain)); break; case QUA: gain *= gain; break; case CUB: gain = CUBE(gain); break; case SQU: gain = sqrt(gain); break; case CBR: gain = cbrt(gain); break; case DESE: gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2; break; case DESI: gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2; break; case LOSI: { const double a = 1. / (1. - 0.787) - 1; double A = 1. / (1.0 + exp(0 -((gain-0.5) * a * 2.0))); double B = 1. / (1.0 + exp(a)); double C = 1. / (1.0 + exp(0-a)); gain = (A - B) / (C - B); } break; case SINC: gain = gain >= 1.0 ? 1.0 : sin(M_PI * (1.0 - gain)) / (M_PI * (1.0 - gain)); break; case ISINC: gain = gain <= 0.0 ? 0.0 : 1.0 - sin(M_PI * gain) / (M_PI * gain); break; case NONE: gain = 1.0; break; } return silence + (unity - silence) * 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, int64_t range,int curve,\ double silence, double unity) \ { \ int i, c; \ \ for (i = 0; i < nb_samples; i++) { \ double gain = fade_gain(curve, start + i * dir,range,silence,unity);\ 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, int64_t range, int curve, \ double silence, double unity) \ { \ 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,silence,unity);\ 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) #define SCALE_PLANAR(name, type) \ static void scale_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \ int nb_samples, int channels, \ double gain) \ { \ int i, c; \ \ for (i = 0; i < nb_samples; i++) { \ for (c = 0; c < channels; c++) { \ type *d = (type *)dst[c]; \ const type *s = (type *)src[c]; \ \ d[i] = s[i] * gain; \ } \ } \ } #define SCALE(name, type) \ static void scale_samples_## name (uint8_t **dst, uint8_t * const *src, \ int nb_samples, int channels, double gain)\ { \ type *d = (type *)dst[0]; \ const type *s = (type *)src[0]; \ int i, c, k = 0; \ \ for (i = 0; i < nb_samples; i++) { \ for (c = 0; c < channels; c++, k++) \ d[k] = s[k] * gain; \ } \ } SCALE_PLANAR(dbl, double) SCALE_PLANAR(flt, float) SCALE_PLANAR(s16, int16_t) SCALE_PLANAR(s32, int32_t) SCALE(dbl, double) SCALE(flt, float) SCALE(s16, int16_t) SCALE(s32, int32_t) static int config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AudioFadeContext *s = ctx->priv; switch (outlink->format) { case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; s->scale_samples = scale_samples_dbl; break; case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; s->scale_samples = scale_samples_dblp; break; case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; s->scale_samples = scale_samples_flt; break; case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; s->scale_samples = scale_samples_fltp; break; case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; s->scale_samples = scale_samples_s16; break; case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; s->scale_samples = scale_samples_s16p; break; case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; s->scale_samples = scale_samples_s32; break; case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; s->scale_samples = scale_samples_s32p; break; } if (s->duration) s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE); s->duration = 0; if (s->start_time) s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE); s->start_time = 0; return 0; } #if CONFIG_AFADE_FILTER static const AVOption afade_options[] = { { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, "type" }, { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, TFLAGS, "type" }, { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, TFLAGS, "type" }, { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, TFLAGS, "type" }, { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS }, { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, TFLAGS }, { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS }, { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 44100}, 1, INT64_MAX, TFLAGS }, { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS }, { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS }, { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS }, { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, INT64_MAX, TFLAGS }, { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, "curve" }, { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, TFLAGS, "curve" }, { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, TFLAGS, "curve" }, { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, TFLAGS, "curve" }, { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, TFLAGS, "curve" }, { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, TFLAGS, "curve" }, { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, TFLAGS, "curve" }, { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, TFLAGS, "curve" }, { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, TFLAGS, "curve" }, { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, TFLAGS, "curve" }, { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, TFLAGS, "curve" }, { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, TFLAGS, "curve" }, { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, TFLAGS, "curve" }, { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, TFLAGS, "curve" }, { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, TFLAGS, "curve" }, { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, TFLAGS, "curve" }, { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, TFLAGS, "curve" }, { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, TFLAGS, "curve" }, { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, TFLAGS, "curve" }, { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, TFLAGS, "curve" }, { "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, TFLAGS, "curve" }, { "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, TFLAGS, "curve" }, { "silence", "set the silence gain", OFFSET(silence), AV_OPT_TYPE_DOUBLE, {.dbl = 0 }, 0, 1, TFLAGS }, { "unity", "set the unity gain", OFFSET(unity), AV_OPT_TYPE_DOUBLE, {.dbl = 1 }, 0, 1, TFLAGS }, { 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 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->unity == 1.0 && ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) || ( s->type && (cur_sample + 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(outlink, 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))) { if (s->silence == 0.) { av_samples_set_silence(out_buf->extended_data, 0, nb_samples, out_buf->ch_layout.nb_channels, out_buf->format); } else { s->scale_samples(out_buf->extended_data, buf->extended_data, nb_samples, buf->ch_layout.nb_channels, s->silence); } } else if (( s->type && (cur_sample + nb_samples < s->start_sample)) || (!s->type && (s->start_sample + s->nb_samples < cur_sample))) { s->scale_samples(out_buf->extended_data, buf->extended_data, nb_samples, buf->ch_layout.nb_channels, s->unity); } 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, buf->ch_layout.nb_channels, s->type ? -1 : 1, start, s->nb_samples, s->curve, s->silence, s->unity); } if (buf != out_buf) av_frame_free(&buf); return ff_filter_frame(outlink, out_buf); } static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags) { int ret; ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); if (ret < 0) return ret; return config_output(ctx->outputs[0]); } static const AVFilterPad avfilter_af_afade_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .filter_frame = filter_frame, }, }; static const AVFilterPad avfilter_af_afade_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }, }; const AVFilter ff_af_afade = { .name = "afade", .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."), .priv_size = sizeof(AudioFadeContext), .init = init, FILTER_INPUTS(avfilter_af_afade_inputs), FILTER_OUTPUTS(avfilter_af_afade_outputs), FILTER_SAMPLEFMTS_ARRAY(sample_fmts), .priv_class = &afade_class, .process_command = process_command, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, }; #endif /* CONFIG_AFADE_FILTER */ #if CONFIG_ACROSSFADE_FILTER static const AVOption acrossfade_options[] = { { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS }, { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS }, { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS }, { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0 }, 0, 60000000, FLAGS }, { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS }, { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS }, { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" }, { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" }, { "nofade", "no fade; keep audio as-is", 0, AV_OPT_TYPE_CONST, {.i64 = NONE }, 0, 0, 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" }, { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 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" }, { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" }, { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" }, { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" }, { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" }, { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" }, { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" }, { "losi", "logistic sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = LOSI }, 0, 0, FLAGS, "curve" }, { "sinc", "sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = SINC }, 0, 0, FLAGS, "curve" }, { "isinc", "inverted sine cardinal function", 0, AV_OPT_TYPE_CONST, {.i64 = ISINC}, 0, 0, FLAGS, "curve" }, { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" }, { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, NONE, NB_CURVES - 1, FLAGS, "curve" }, { NULL } }; AVFILTER_DEFINE_CLASS(acrossfade); #define CROSSFADE_PLANAR(name, type) \ static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \ uint8_t * const *cf1, \ int nb_samples, int channels, \ int curve0, int curve1) \ { \ int i, c; \ \ for (i = 0; i < nb_samples; i++) { \ double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples,0.,1.);\ double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \ for (c = 0; c < channels; c++) { \ type *d = (type *)dst[c]; \ const type *s0 = (type *)cf0[c]; \ const type *s1 = (type *)cf1[c]; \ \ d[i] = s0[i] * gain0 + s1[i] * gain1; \ } \ } \ } #define CROSSFADE(name, type) \ static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \ uint8_t * const *cf1, \ int nb_samples, int channels, \ int curve0, int curve1) \ { \ type *d = (type *)dst[0]; \ const type *s0 = (type *)cf0[0]; \ const type *s1 = (type *)cf1[0]; \ int i, c, k = 0; \ \ for (i = 0; i < nb_samples; i++) { \ double gain0 = fade_gain(curve0, nb_samples - 1-i,nb_samples,0.,1.);\ double gain1 = fade_gain(curve1, i, nb_samples, 0., 1.); \ for (c = 0; c < channels; c++, k++) \ d[k] = s0[k] * gain0 + s1[k] * gain1; \ } \ } CROSSFADE_PLANAR(dbl, double) CROSSFADE_PLANAR(flt, float) CROSSFADE_PLANAR(s16, int16_t) CROSSFADE_PLANAR(s32, int32_t) CROSSFADE(dbl, double) CROSSFADE(flt, float) CROSSFADE(s16, int16_t) CROSSFADE(s32, int32_t) static int activate(AVFilterContext *ctx) { AudioFadeContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; AVFrame *in = NULL, *out, *cf[2] = { NULL }; int ret = 0, nb_samples, status; int64_t pts; FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx); if (s->crossfade_is_over) { ret = ff_inlink_consume_frame(ctx->inputs[1], &in); if (ret > 0) { in->pts = s->pts; s->pts += av_rescale_q(in->nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); return ff_filter_frame(outlink, in); } else if (ret < 0) { return ret; } else if (ff_inlink_acknowledge_status(ctx->inputs[1], &status, &pts)) { ff_outlink_set_status(ctx->outputs[0], status, pts); return 0; } else if (!ret) { if (ff_outlink_frame_wanted(ctx->outputs[0])) { ff_inlink_request_frame(ctx->inputs[1]); return 0; } } } nb_samples = ff_inlink_queued_samples(ctx->inputs[0]); if (nb_samples > s->nb_samples) { nb_samples -= s->nb_samples; ret = ff_inlink_consume_samples(ctx->inputs[0], nb_samples, nb_samples, &in); if (ret < 0) return ret; in->pts = s->pts; s->pts += av_rescale_q(in->nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); return ff_filter_frame(outlink, in); } else if (s->cf0_eof && nb_samples >= s->nb_samples && ff_inlink_queued_samples(ctx->inputs[1]) >= s->nb_samples) { if (s->overlap) { out = ff_get_audio_buffer(outlink, s->nb_samples); if (!out) return AVERROR(ENOMEM); ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]); if (ret < 0) { av_frame_free(&out); return ret; } ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]); if (ret < 0) { av_frame_free(&out); return ret; } s->crossfade_samples(out->extended_data, cf[0]->extended_data, cf[1]->extended_data, s->nb_samples, out->ch_layout.nb_channels, s->curve, s->curve2); out->pts = s->pts; s->pts += av_rescale_q(s->nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); s->crossfade_is_over = 1; av_frame_free(&cf[0]); av_frame_free(&cf[1]); return ff_filter_frame(outlink, out); } else { out = ff_get_audio_buffer(outlink, s->nb_samples); if (!out) return AVERROR(ENOMEM); ret = ff_inlink_consume_samples(ctx->inputs[0], s->nb_samples, s->nb_samples, &cf[0]); if (ret < 0) { av_frame_free(&out); return ret; } s->fade_samples(out->extended_data, cf[0]->extended_data, s->nb_samples, outlink->ch_layout.nb_channels, -1, s->nb_samples - 1, s->nb_samples, s->curve, 0., 1.); out->pts = s->pts; s->pts += av_rescale_q(s->nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); av_frame_free(&cf[0]); ret = ff_filter_frame(outlink, out); if (ret < 0) return ret; out = ff_get_audio_buffer(outlink, s->nb_samples); if (!out) return AVERROR(ENOMEM); ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_samples, s->nb_samples, &cf[1]); if (ret < 0) { av_frame_free(&out); return ret; } s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples, outlink->ch_layout.nb_channels, 1, 0, s->nb_samples, s->curve2, 0., 1.); out->pts = s->pts; s->pts += av_rescale_q(s->nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base); s->crossfade_is_over = 1; av_frame_free(&cf[1]); return ff_filter_frame(outlink, out); } } else if (ff_outlink_frame_wanted(ctx->outputs[0])) { if (!s->cf0_eof && ff_outlink_get_status(ctx->inputs[0])) { s->cf0_eof = 1; } if (ff_outlink_get_status(ctx->inputs[1])) { ff_outlink_set_status(ctx->outputs[0], AVERROR_EOF, AV_NOPTS_VALUE); return 0; } if (!s->cf0_eof) ff_inlink_request_frame(ctx->inputs[0]); else ff_inlink_request_frame(ctx->inputs[1]); return 0; } return ret; } static int acrossfade_config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AudioFadeContext *s = ctx->priv; outlink->time_base = ctx->inputs[0]->time_base; switch (outlink->format) { case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break; case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break; case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break; case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break; case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break; case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break; case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break; case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break; } config_output(outlink); return 0; } static const AVFilterPad avfilter_af_acrossfade_inputs[] = { { .name = "crossfade0", .type = AVMEDIA_TYPE_AUDIO, }, { .name = "crossfade1", .type = AVMEDIA_TYPE_AUDIO, }, }; static const AVFilterPad avfilter_af_acrossfade_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = acrossfade_config_output, }, }; const AVFilter ff_af_acrossfade = { .name = "acrossfade", .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."), .priv_size = sizeof(AudioFadeContext), .activate = activate, .priv_class = &acrossfade_class, FILTER_INPUTS(avfilter_af_acrossfade_inputs), FILTER_OUTPUTS(avfilter_af_acrossfade_outputs), FILTER_SAMPLEFMTS_ARRAY(sample_fmts), }; #endif /* CONFIG_ACROSSFADE_FILTER */