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avfilter/af_biquads: add option for block based linear phase processing

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This commit is contained in:
Paul B Mahol 2022-05-09 01:33:50 +02:00
parent fe57904a71
commit 1309867022
2 changed files with 186 additions and 4 deletions
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56
doc/filters.texi
View File

@ -3500,6 +3500,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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
@ -3589,6 +3596,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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
@ -3688,6 +3702,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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
@ -3772,6 +3793,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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
@section bs2b
@ -4566,6 +4594,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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 Examples
@ -5069,6 +5104,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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
@ -5421,6 +5463,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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 Examples
@ -6659,6 +6708,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
@item block_size, b
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

134
libavfilter/af_biquads.c
View File

@ -70,6 +70,7 @@
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
enum FilterType {
@ -109,6 +110,8 @@ enum TransformType {
typedef struct ChanCache {
double i1, i2;
double o1, o2;
double ri1, ri2;
double ro1, ro2;
int clippings;
} ChanCache;
@ -121,6 +124,7 @@ typedef struct BiquadsContext {
int csg;
int transform_type;
int precision;
int block_samples;
int bypass;
@ -139,6 +143,8 @@ typedef struct BiquadsContext {
double oa0, oa1, oa2;
double ob0, ob1, ob2;
AVFrame *block[3];
ChanCache *cache;
int block_align;
@ -782,6 +788,14 @@ static int config_filter(AVFilterLink *outlink, int reset)
if (reset)
memset(s->cache, 0, sizeof(ChanCache) * inlink->ch_layout.nb_channels);
if (reset && s->block_samples > 0) {
for (int i = 0; i < 3; i++) {
s->block[i] = ff_get_audio_buffer(outlink, s->block_samples * 2);
if (!s->block[i])
return AVERROR(ENOMEM);
}
}
switch (s->transform_type) {
case DI:
switch (inlink->format) {
@ -908,6 +922,41 @@ typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
static void reverse_samples(AVFrame *out, AVFrame *in, int p,
int oo, int io, int nb_samples)
{
switch (out->format) {
case AV_SAMPLE_FMT_S16P: {
const int16_t *src = ((const int16_t *)out->extended_data[p]) + io;
int16_t *dst = ((int16_t *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
case AV_SAMPLE_FMT_S32P: {
const int32_t *src = ((const int32_t *)out->extended_data[p]) + io;
int32_t *dst = ((int32_t *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
case AV_SAMPLE_FMT_FLTP: {
const float *src = ((const float *)in->extended_data[p]) + io;
float *dst = ((float *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
case AV_SAMPLE_FMT_DBLP: {
const double *src = ((const double *)in->extended_data[p]) + io;
double *dst = ((double *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
}
}
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AVFilterLink *inlink = ctx->inputs[0];
@ -930,9 +979,37 @@ static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_job
continue;
}
s->filter(s, buf->extended_data[ch], out_buf->extended_data[ch], buf->nb_samples,
&s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2,
s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
if (!s->block_samples) {
s->filter(s, buf->extended_data[ch], out_buf->extended_data[ch], buf->nb_samples,
&s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2,
s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
} else {
memcpy(s->block[0]->extended_data[ch] + s->block_align * s->block_samples, buf->extended_data[ch],
buf->nb_samples * s->block_align);
s->filter(s, s->block[0]->extended_data[ch], s->block[1]->extended_data[ch], s->block_samples,
&s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2,
s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
s->cache[ch].ri1 = s->cache[ch].i1;
s->cache[ch].ri2 = s->cache[ch].i2;
s->cache[ch].ro1 = s->cache[ch].o1;
s->cache[ch].ro2 = s->cache[ch].o2;
s->filter(s, s->block[0]->extended_data[ch] + s->block_samples * s->block_align,
s->block[1]->extended_data[ch] + s->block_samples * s->block_align,
s->block_samples,
&s->cache[ch].ri1, &s->cache[ch].ri2, &s->cache[ch].ro1, &s->cache[ch].ro2,
s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
reverse_samples(s->block[2], s->block[1], ch, 0, 0, s->block_samples * 2);
s->cache[ch].ri1 = 0.;
s->cache[ch].ri2 = 0.;
s->cache[ch].ro1 = 0.;
s->cache[ch].ro2 = 0.;
s->filter(s, s->block[2]->extended_data[ch], s->block[2]->extended_data[ch], s->block[2]->nb_samples,
&s->cache[ch].ri1, &s->cache[ch].ri2, &s->cache[ch].ro1, &s->cache[ch].ro2,
s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
reverse_samples(out_buf, s->block[2], ch, 0, s->block_samples, out_buf->nb_samples);
memmove(s->block[0]->extended_data[ch], s->block[0]->extended_data[ch] + s->block_align * s->block_samples,
s->block_samples * s->block_align);
}
}
return 0;
@ -988,6 +1065,37 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
return ff_filter_frame(outlink, out_buf);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
BiquadsContext *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)
{
@ -1005,6 +1113,8 @@ static av_cold void uninit(AVFilterContext *ctx)
{
BiquadsContext *s = ctx->priv;
for (int i = 0; i < 3; i++)
av_frame_free(&s->block[i]);
av_freep(&s->cache);
av_channel_layout_uninit(&s->ch_layout);
}
@ -1013,7 +1123,6 @@ static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
@ -1043,6 +1152,7 @@ const AVFilter ff_af_##name_ = { \
.priv_class = &priv_class_##_class, \
.priv_size = sizeof(BiquadsContext), \
.init = name_##_init, \
.activate = activate, \
.uninit = uninit, \
FILTER_INPUTS(inputs), \
FILTER_OUTPUTS(outputs), \
@ -1091,6 +1201,8 @@ static const AVOption equalizer_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1134,6 +1246,8 @@ static const AVOption bass_lowshelf_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1184,6 +1298,8 @@ static const AVOption treble_highshelf_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1233,6 +1349,8 @@ static const AVOption bandpass_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1272,6 +1390,8 @@ static const AVOption bandreject_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1313,6 +1433,8 @@ static const AVOption lowpass_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1354,6 +1476,8 @@ static const AVOption highpass_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};
@ -1429,6 +1553,8 @@ static const AVOption biquad_options[] = {
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"},
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"},
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"},
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF},
{NULL}
};