[FFmpeg-cvslog] avfilter/af_afade: add options to control unity/silence gains
Paul B Mahol
git at videolan.org
Tue Jan 3 11:20:38 EET 2023
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Tue Jan 3 10:20:52 2023 +0100| [c94988a7814a8268a4732725472bed8b3aa1ba74] | committer: Paul B Mahol
avfilter/af_afade: add options to control unity/silence gains
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c94988a7814a8268a4732725472bed8b3aa1ba74
---
doc/filters.texi | 8 +++
libavfilter/af_afade.c | 130 +++++++++++++++++++++++++++++++++++++++----------
2 files changed, 112 insertions(+), 26 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 9b866de5ae..d0ec0a2181 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1380,6 +1380,14 @@ select inverted sine cardinal function
@item nofade
no fade applied
@end table
+
+ at item silence
+Set the initial gain for fade-in or final gain for fade-out.
+Default value is @code{0.0}.
+
+ at item unity
+Set the initial gain for fade-out or final gain for fade-in.
+Default value is @code{1.0}.
@end table
@subsection Commands
diff --git a/libavfilter/af_afade.c b/libavfilter/af_afade.c
index 0445bb729f..48938384ff 100644
--- a/libavfilter/af_afade.c
+++ b/libavfilter/af_afade.c
@@ -39,6 +39,8 @@ typedef struct AudioFadeContext {
int64_t start_sample;
int64_t duration;
int64_t start_time;
+ double silence;
+ double unity;
int overlap;
int cf0_eof;
int crossfade_is_over;
@@ -46,7 +48,10 @@ typedef struct AudioFadeContext {
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);
+ 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,
@@ -67,7 +72,7 @@ enum CurveType { NONE = -1, TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR
AV_SAMPLE_FMT_NONE
};
-static double fade_gain(int curve, int64_t index, int64_t range)
+static double fade_gain(int curve, int64_t index, int64_t range, double silence, double unity)
{
#define CUBE(a) ((a)*(a)*(a))
double gain;
@@ -142,18 +147,19 @@ static double fade_gain(int curve, int64_t index, int64_t range)
break;
}
- return gain;
+ 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) \
+ 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); \
+ 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]; \
@@ -166,14 +172,15 @@ static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
#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) \
+ 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); \
+ double gain = fade_gain(curve, start + i * dir,range,silence,unity);\
for (c = 0; c < channels; c++, k++) \
d[k] = s[k] * gain; \
} \
@@ -189,20 +196,77 @@ 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; 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;
+ 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)
@@ -252,6 +316,8 @@ static const AVOption afade_options[] = {
{ "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 }
};
@@ -275,8 +341,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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 + nb_samples < s->start_sample)))
+ 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)) {
@@ -290,8 +357,19 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *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,
- out_buf->ch_layout.nb_channels, out_buf->format);
+ 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;
@@ -303,7 +381,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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->nb_samples, s->curve, s->silence, s->unity);
}
if (buf != out_buf)
@@ -402,8 +480,8 @@ static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
int i, c; \
\
for (i = 0; i < nb_samples; i++) { \
- double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
- double gain1 = fade_gain(curve1, i, nb_samples); \
+ 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]; \
@@ -426,8 +504,8 @@ static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
int i, c, k = 0; \
\
for (i = 0; i < nb_samples; i++) { \
- double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
- double gain1 = fade_gain(curve1, i, nb_samples); \
+ 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; \
} \
@@ -525,7 +603,7 @@ static int activate(AVFilterContext *ctx)
}
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);
+ 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);
@@ -545,7 +623,7 @@ static int activate(AVFilterContext *ctx)
}
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);
+ 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);
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