[FFmpeg-cvslog] avfilter/af_acrossover: add per output band gain
Paul B Mahol
git at videolan.org
Wed Dec 2 14:57:02 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Dec 2 13:48:32 2020 +0100| [0c8a0d3a56b62c4123a76add0ae615466f5ff7da] | committer: Paul B Mahol
avfilter/af_acrossover: add per output band gain
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0c8a0d3a56b62c4123a76add0ae615466f5ff7da
---
doc/filters.texi | 3 +++
libavfilter/af_acrossover.c | 60 +++++++++++++++++++++++++++++++++++++++------
2 files changed, 55 insertions(+), 8 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index f7b8f92629..a4662e78f2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -554,6 +554,9 @@ Default is @var{4th}.
@item level
Set input gain level. Allowed range is from 0 to 1. Default value is 1.
+
+ at item gains
+Set output gain for each band. Default value is 1 for all bands.
@end table
@subsection Examples
diff --git a/libavfilter/af_acrossover.c b/libavfilter/af_acrossover.c
index 60bb2330f0..03a7ae1119 100644
--- a/libavfilter/af_acrossover.c
+++ b/libavfilter/af_acrossover.c
@@ -54,6 +54,7 @@ typedef struct AudioCrossoverContext {
const AVClass *class;
char *splits_str;
+ char *gains_str;
int order_opt;
float level_in;
@@ -64,6 +65,8 @@ typedef struct AudioCrossoverContext {
int nb_splits;
float splits[MAX_SPLITS];
+ float gains[MAX_BANDS];
+
BiquadCoeffs lp[MAX_BANDS][20];
BiquadCoeffs hp[MAX_BANDS][20];
BiquadCoeffs ap[MAX_BANDS][20];
@@ -95,11 +98,47 @@ static const AVOption acrossover_options[] = {
{ "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
{ "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" },
{ "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
+ { "gain", "set output bands gain", OFFSET(gains_str), AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrossover);
+static int parse_gains(AVFilterContext *ctx)
+{
+ AudioCrossoverContext *s = ctx->priv;
+ char *p, *arg, *saveptr = NULL;
+ int i, ret = 0;
+
+ saveptr = NULL;
+ p = s->gains_str;
+ for (i = 0; i < MAX_BANDS; i++) {
+ float gain;
+ char c[3] = { 0 };
+
+ if (!(arg = av_strtok(p, " |", &saveptr)))
+ break;
+
+ p = NULL;
+
+ if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);
+ ret = AVERROR(EINVAL);
+ break;
+ }
+
+ if (c[0] == 'd' && c[1] == 'B')
+ s->gains[i] = expf(gain * M_LN10 / 20.f);
+ else
+ s->gains[i] = gain;
+ }
+
+ for (; i < MAX_BANDS; i++)
+ s->gains[i] = 1.f;
+
+ return ret;
+}
+
static av_cold int init(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
@@ -138,6 +177,10 @@ static av_cold int init(AVFilterContext *ctx)
s->nb_splits = i;
+ ret = parse_gains(ctx);
+ if (ret < 0)
+ return ret;
+
for (i = 0; i <= s->nb_splits; i++) {
AVFilterPad pad = { 0 };
char *name;
@@ -349,6 +392,7 @@ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, i
const int end = (in->channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = in->nb_samples; \
const int nb_outs = ctx->nb_outputs; \
+ const int first_order = s->first_order; \
\
for (int ch = start; ch < end; ch++) { \
const type *src = (const type *)in->extended_data[ch]; \
@@ -378,7 +422,7 @@ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, i
} \
\
for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \
- if (s->first_order) { \
+ if (first_order) { \
const type *asrc = (const type *)frames[band]->extended_data[ch]; \
type *dst = (type *)frames[band]->extended_data[ch]; \
type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \
@@ -387,7 +431,7 @@ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, i
biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
} \
\
- for (int f = s->first_order; f < s->ap_filter_count; f++) { \
+ for (int f = first_order; f < s->ap_filter_count; f++) { \
const type *asrc = (const type *)frames[band]->extended_data[ch]; \
type *dst = (type *)frames[band]->extended_data[ch]; \
type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\
@@ -398,12 +442,12 @@ static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, i
} \
} \
\
- for (int band = 0; band < nb_outs && s->first_order; band++) { \
- if (band & 1) { \
- type *dst = (type *)frames[band]->extended_data[ch]; \
- s->fdsp->vector_## ff ##mul_scalar(dst, dst, -one, \
- FFALIGN(nb_samples, sizeof(type))); \
- } \
+ for (int band = 0; band < nb_outs; band++) { \
+ const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \
+ type *dst = (type *)frames[band]->extended_data[ch]; \
+ \
+ s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \
+ FFALIGN(nb_samples, sizeof(type))); \
} \
} \
\
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