[FFmpeg-cvslog] avfilter/af_biquads: add option for block based linear phase processing
Paul B Mahol
git at videolan.org
Mon May 9 23:21:20 EEST 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon May 9 01:33:50 2022 +0200| [1309867022a5edf74e80ee6114049d7d053296fc] | committer: Paul B Mahol
avfilter/af_biquads: add option for block based linear phase processing
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1309867022a5edf74e80ee6114049d7d053296fc
---
doc/filters.texi | 56 ++++++++++++++++++++
libavfilter/af_biquads.c | 134 +++++++++++++++++++++++++++++++++++++++++++++--
2 files changed, 186 insertions(+), 4 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index dc8bb54994..468c277798 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3500,6 +3500,13 @@ Always use float 32-bit.
@item f64
Always use float 64-bit.
@end table
+
+ at 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
+
+ at 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
+
+ at 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
+
+ at 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
+
+ at 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
+
+ at 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
+
+ at 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
+
+ at 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
diff --git a/libavfilter/af_biquads.c b/libavfilter/af_biquads.c
index 8caf169d50..2ec32e915d 100644
--- a/libavfilter/af_biquads.c
+++ b/libavfilter/af_biquads.c
@@ -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}
};
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