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authorPaul B Mahol <onemda@gmail.com>2013-01-17 21:53:27 +0000
committerPaul B Mahol <onemda@gmail.com>2013-01-21 15:20:42 +0000
commit6ea8a830e8f1eee465174c479840b18b4963d43d (patch)
tree5ffc84350ba346a2599de549c16d92125d539cc2 /libavfilter/af_afade.c
parent5f61e09a8f28af45d7db1f5e7767d14322255f49 (diff)
afade filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Diffstat (limited to 'libavfilter/af_afade.c')
-rw-r--r--libavfilter/af_afade.c307
1 files changed, 307 insertions, 0 deletions
diff --git a/libavfilter/af_afade.c b/libavfilter/af_afade.c
new file mode 100644
index 0000000000..00a05e2c1e
--- /dev/null
+++ b/libavfilter/af_afade.c
@@ -0,0 +1,307 @@
+/*
+ * 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;
+ double duration;
+ double 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", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
+ { "out", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
+ { "start_sample", "set expression of sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
+ { "ss", "set expression of sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
+ { "nb_samples", "set expression for fade duration in samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
+ { "ns", "set expression for fade duration in samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
+ { "start_time", "set expression of second to start fading", OFFSET(start_time), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 7*24*60*60,FLAGS },
+ { "st", "set expression of second to start fading", OFFSET(start_time), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 7*24*60*60,FLAGS },
+ { "duration", "set expression for fade duration in seconds", OFFSET(duration), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 24*60*60, FLAGS },
+ { "d", "set expression for fade duration in seconds", OFFSET(duration), AV_OPT_TYPE_DOUBLE, {.dbl = 0. }, 0, 24*60*60, FLAGS },
+ { "curve", "set expression for fade curve", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, TRI, CBR, FLAGS, "curve" },
+ { "c", "set expression for fade curve", 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, const char *args)
+{
+ AudioFadeContext *afade = ctx->priv;
+ int ret;
+
+ afade->class = &afade_class;
+ av_opt_set_defaults(afade);
+
+ if ((ret = av_set_options_string(afade, args, "=", ":")) < 0)
+ return ret;
+
+ if (INT64_MAX - afade->nb_samples < afade->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_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFadeContext *afade = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_DBL: afade->fade_samples = fade_samples_dbl; break;
+ case AV_SAMPLE_FMT_DBLP: afade->fade_samples = fade_samples_dblp; break;
+ case AV_SAMPLE_FMT_FLT: afade->fade_samples = fade_samples_flt; break;
+ case AV_SAMPLE_FMT_FLTP: afade->fade_samples = fade_samples_fltp; break;
+ case AV_SAMPLE_FMT_S16: afade->fade_samples = fade_samples_s16; break;
+ case AV_SAMPLE_FMT_S16P: afade->fade_samples = fade_samples_s16p; break;
+ case AV_SAMPLE_FMT_S32: afade->fade_samples = fade_samples_s32; break;
+ case AV_SAMPLE_FMT_S32P: afade->fade_samples = fade_samples_s32p; break;
+ }
+
+ if (afade->duration)
+ afade->nb_samples = afade->duration * inlink->sample_rate;
+ if (afade->start_time)
+ afade->start_sample = afade->start_time * inlink->sample_rate;
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ AudioFadeContext *afade = inlink->dst->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ int nb_samples = buf->audio->nb_samples;
+ AVFilterBufferRef *out_buf;
+ int64_t cur_sample = av_rescale_q(buf->pts, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+
+ if ((!afade->type && (afade->start_sample + afade->nb_samples < cur_sample)) ||
+ ( afade->type && (cur_sample + afade->nb_samples < afade->start_sample)))
+ return ff_filter_frame(outlink, buf);
+
+ if (buf->perms & AV_PERM_WRITE) {
+ out_buf = buf;
+ } else {
+ out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
+ if (!out_buf)
+ return AVERROR(ENOMEM);
+ out_buf->pts = buf->pts;
+ }
+
+ if ((!afade->type && (cur_sample + nb_samples < afade->start_sample)) ||
+ ( afade->type && (afade->start_sample + afade->nb_samples < cur_sample))) {
+ av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
+ out_buf->audio->channels, out_buf->format);
+ } else {
+ int64_t start;
+
+ if (!afade->type)
+ start = cur_sample - afade->start_sample;
+ else
+ start = afade->start_sample + afade->nb_samples - cur_sample;
+
+ afade->fade_samples(out_buf->extended_data, buf->extended_data,
+ nb_samples, buf->audio->channels,
+ afade->type ? -1 : 1, start,
+ afade->nb_samples, afade->curve);
+ }
+
+ if (buf != out_buf)
+ avfilter_unref_buffer(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,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_afade_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+AVFilter avfilter_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,
+};