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avfilter/af_crossfeed: add option for linear phase filtering 

But at expense of additional delay.
This commit is contained in:
Paul B Mahol 2022-05-09 21:59:03 +02:00
parent 1309867022
commit 5ccd8f492b
2 changed files with 167 additions and 14 deletions
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7
doc/filters.texi
View File

@ -4205,6 +4205,13 @@ Set input gain. Default is 0.9.
@item level_out
Set output gain. Default is 1.
@item block_size
Set block size used for reverse IIR processing. If this value is set to high enough
value (higher than impulse response length truncated when reaches near zero values) filtering
will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
@end table
@subsection Commands

174
libavfilter/af_crossfeed.c
View File

@ -21,6 +21,7 @@
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "formats.h"
typedef struct CrossfeedContext {
@ -31,11 +32,16 @@ typedef struct CrossfeedContext {
double slope;
double level_in;
double level_out;
int block_samples;
int block_size;
double a0, a1, a2;
double b0, b1, b2;
double w1, w2;
double *mid;
double *side[3];
} CrossfeedContext;
static int query_formats(AVFilterContext *ctx)
@ -77,9 +83,49 @@ static int config_input(AVFilterLink *inlink)
s->b1 /= s->a0;
s->b2 /= s->a0;
if (s->block_samples == 0 && s->block_size > 0) {
s->block_samples = s->block_size;
s->mid = av_calloc(s->block_samples * 2, sizeof(*s->mid));
for (int i = 0; i < 3; i++) {
s->side[i] = av_calloc(s->block_samples * 2, sizeof(*s->side[0]));
if (!s->side[i])
return AVERROR(ENOMEM);
}
}
return 0;
}
static void reverse_samples(double *dst, const double *src,
int nb_samples)
{
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
static void filter_samples(double *dst, const double *src,
int nb_samples,
double b0, double b1, double b2,
double a1, double a2,
double *sw1, double *sw2)
{
double w1 = *sw1;
double w2 = *sw2;
for (int n = 0; n < nb_samples; n++) {
double side = src[n];
double oside = side * b0 + w1;
w1 = b1 * side + w2 + a1 * oside;
w2 = b2 * side + a2 * oside;
dst[n] = oside;
}
*sw1 = w1;
*sw2 = w2;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
@ -95,7 +141,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
const double a2 = -s->a2;
AVFrame *out;
double *dst;
int n;
if (av_frame_is_writable(in)) {
out = in;
@ -109,21 +154,79 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
}
dst = (double *)out->data[0];
for (n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
double mid = (src[0] + src[1]) * level_in * .5;
double side = (src[0] - src[1]) * level_in * .5;
double oside = side * b0 + s->w1;
if (s->block_samples == 0) {
double w1 = s->w1;
double w2 = s->w2;
s->w1 = b1 * side + s->w2 + a1 * oside;
s->w2 = b2 * side + a2 * oside;
for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
double mid = (src[0] + src[1]) * level_in * .5;
double side = (src[0] - src[1]) * level_in * .5;
double oside = side * b0 + w1;
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = (mid + oside) * level_out;
dst[1] = (mid - oside) * level_out;
w1 = b1 * side + w2 + a1 * oside;
w2 = b2 * side + a2 * oside;
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = (mid + oside) * level_out;
dst[1] = (mid - oside) * level_out;
}
}
s->w1 = w1;
s->w2 = w2;
} else {
double *mdst = s->mid + s->block_samples;
double *sdst = s->side[0] + s->block_samples;
double *ssrc = s->side[0];
double *msrc = s->mid;
double w1 = s->w1;
double w2 = s->w2;
for (int n = 0; n < out->nb_samples; n++, src += 2) {
mdst[n] = (src[0] + src[1]) * level_in * .5;
sdst[n] = (src[0] - src[1]) * level_in * .5;
}
sdst = s->side[1];
filter_samples(sdst, ssrc, s->block_samples,
b0, b1, b2, a1, a2,
&w1, &w2);
s->w1 = w1;
s->w2 = w2;
ssrc = s->side[0] + s->block_samples;
sdst = s->side[1] + s->block_samples;
filter_samples(sdst, ssrc, s->block_samples,
b0, b1, b2, a1, a2,
&w1, &w2);
reverse_samples(s->side[2], s->side[1], s->block_samples * 2);
w1 = w2 = 0.;
filter_samples(s->side[2], s->side[2], s->block_samples * 2,
b0, b1, b2, a1, a2,
&w1, &w2);
reverse_samples(s->side[1], s->side[2] + s->block_samples, s->block_samples);
src = (const double *)in->data[0];
ssrc = s->side[1];
for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = (msrc[n] + ssrc[n]) * level_out;
dst[1] = (msrc[n] - ssrc[n]) * level_out;
}
}
memmove(s->mid, s->mid + s->block_samples,
s->block_samples * sizeof(*s->mid));
memmove(s->side[0], s->side[0] + s->block_samples,
s->block_samples * sizeof(*s->side[0]));
}
if (out != in)
@ -131,6 +234,37 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
return ff_filter_frame(outlink, out);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
CrossfeedContext *s = ctx->priv;
AVFrame *in = NULL;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (s->block_samples > 0) {
ret = ff_inlink_consume_samples(inlink, s->block_samples, s->block_samples, &in);
} else {
ret = ff_inlink_consume_frame(inlink, &in);
}
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, in);
if (s->block_samples > 0 && ff_inlink_queued_samples(inlink) >= s->block_samples) {
ff_filter_set_ready(ctx, 10);
return 0;
}
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
@ -143,8 +277,18 @@ static int process_command(AVFilterContext *ctx, const char *cmd, const char *ar
return config_input(ctx->inputs[0]);
}
static av_cold void uninit(AVFilterContext *ctx)
{
CrossfeedContext *s = ctx->priv;
av_freep(&s->mid);
for (int i = 0; i < 3; i++)
av_freep(&s->side[i]);
}
#define OFFSET(x) offsetof(CrossfeedContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption crossfeed_options[] = {
{ "strength", "set crossfeed strength", OFFSET(strength), AV_OPT_TYPE_DOUBLE, {.dbl=.2}, 0, 1, FLAGS },
@ -152,6 +296,7 @@ static const AVOption crossfeed_options[] = {
{ "slope", "set curve slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .01, 1, FLAGS },
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=.9}, 0, 1, FLAGS },
{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1.}, 0, 1, FLAGS },
{ "block_size", "set the block size", OFFSET(block_size),AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
{ NULL }
};
@ -161,7 +306,6 @@ static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
@ -178,6 +322,8 @@ const AVFilter ff_af_crossfeed = {
.description = NULL_IF_CONFIG_SMALL("Apply headphone crossfeed filter."),
.priv_size = sizeof(CrossfeedContext),
.priv_class = &crossfeed_class,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),