[FFmpeg-cvslog] avfilter/af_asoftclip: rewrite oversampling
Paul B Mahol
git at videolan.org
Sun Sep 12 13:56:39 EEST 2021
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Sep 12 00:31:13 2021 +0200| [94d4cc24c314393b5df26ab800b3ccf50366bc60] | committer: Paul B Mahol
avfilter/af_asoftclip: rewrite oversampling
Fixes most aliasing issues.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=94d4cc24c314393b5df26ab800b3ccf50366bc60
---
configure | 1 -
libavfilter/af_asoftclip.c | 305 +++++++++++++++++++++++++--------------------
2 files changed, 171 insertions(+), 135 deletions(-)
diff --git a/configure b/configure
index af410a9d11..98987ed186 100755
--- a/configure
+++ b/configure
@@ -3548,7 +3548,6 @@ afir_filter_select="rdft"
ametadata_filter_deps="avformat"
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
-asoftclip_filter_deps="swresample"
asr_filter_deps="pocketsphinx"
ass_filter_deps="libass"
atempo_filter_deps="avcodec"
diff --git a/libavfilter/af_asoftclip.c b/libavfilter/af_asoftclip.c
index a90a4c7eb5..9b3d58747a 100644
--- a/libavfilter/af_asoftclip.c
+++ b/libavfilter/af_asoftclip.c
@@ -21,11 +21,12 @@
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
-#include "libswresample/swresample.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
+#define MAX_OVERSAMPLE 64
+
enum ASoftClipTypes {
ASC_HARD = -1,
ASC_TANH,
@@ -39,6 +40,14 @@ enum ASoftClipTypes {
NB_TYPES,
};
+typedef struct Lowpass {
+ float fb0, fb1, fb2;
+ float fa0, fa1, fa2;
+
+ double db0, db1, db2;
+ double da0, da1, da2;
+} Lowpass;
+
typedef struct ASoftClipContext {
const AVClass *class;
@@ -49,10 +58,8 @@ typedef struct ASoftClipContext {
double output;
double param;
- SwrContext *up_ctx;
- SwrContext *down_ctx;
-
- AVFrame *frame;
+ Lowpass lowpass[MAX_OVERSAMPLE];
+ AVFrame *frame[2];
void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
int nb_samples, int channels, int start, int end);
@@ -60,7 +67,6 @@ typedef struct ASoftClipContext {
#define OFFSET(x) offsetof(ASoftClipContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
-#define F AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption asoftclip_options[] = {
{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" },
@@ -76,7 +82,7 @@ static const AVOption asoftclip_options[] = {
{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
- { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
+ { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, MAX_OVERSAMPLE, A },
{ NULL }
};
@@ -85,8 +91,7 @@ AVFILTER_DEFINE_CLASS(asoftclip);
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
@@ -100,42 +105,103 @@ static int query_formats(AVFilterContext *ctx)
return ff_set_common_all_samplerates(ctx);
}
+static void get_lowpass(Lowpass *s,
+ double frequency,
+ double sample_rate)
+{
+ double w0 = 2 * M_PI * frequency / sample_rate;
+ double alpha = sin(w0) / (2 * 0.8);
+ double factor;
+
+ s->da0 = 1 + alpha;
+ s->da1 = -2 * cos(w0);
+ s->da2 = 1 - alpha;
+ s->db0 = (1 - cos(w0)) / 2;
+ s->db1 = 1 - cos(w0);
+ s->db2 = (1 - cos(w0)) / 2;
+
+ s->da1 /= s->da0;
+ s->da2 /= s->da0;
+ s->db0 /= s->da0;
+ s->db1 /= s->da0;
+ s->db2 /= s->da0;
+ s->da0 /= s->da0;
+
+ factor = (s->da0 + s->da1 + s->da2) / (s->db0 + s->db1 + s->db2);
+ s->db0 *= factor;
+ s->db1 *= factor;
+ s->db2 *= factor;
+
+ s->fa0 = s->da0;
+ s->fa1 = s->da1;
+ s->fa2 = s->da2;
+ s->fb0 = s->db0;
+ s->fb1 = s->db1;
+ s->fb2 = s->db2;
+}
+
+static inline float run_lowpassf(const Lowpass *const s,
+ float src, float *w)
+{
+ float dst;
+
+ dst = src * s->fb0 + w[0];
+ w[0] = s->fb1 * src + w[1] - s->fa1 * dst;
+ w[1] = s->fb2 * src - s->fa2 * dst;
+
+ return dst;
+}
+
static void filter_flt(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
+ const int oversample = s->oversample;
+ const int nb_osamples = nb_samples * oversample;
+ const float scale = oversample > 1 ? oversample * 0.5f : 1.f;
float threshold = s->threshold;
float gain = s->output * threshold;
float factor = 1.f / threshold;
float param = s->param;
for (int c = start; c < end; c++) {
+ float *w = (float *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
const float *src = sptr[c];
float *dst = dptr[c];
+ for (int n = 0; n < nb_samples; n++) {
+ dst[oversample * n] = src[n];
+
+ for (int m = 1; m < oversample; m++)
+ dst[oversample * n + m] = 0.f;
+ }
+
+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
+ dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
+
switch (s->type) {
case ASC_HARD:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
dst[n] *= gain;
}
break;
case ASC_TANH:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = tanhf(src[n] * factor * param);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = tanhf(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
- for (int n = 0; n < nb_samples; n++) {
- float sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ float sample = dst[n] * factor;
if (FFABS(sample) >= 1.5f)
dst[n] = FFSIGN(sample);
@@ -145,22 +211,22 @@ static void filter_flt(ASoftClipContext *s,
}
break;
case ASC_EXP:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
- for (int n = 0; n < nb_samples; n++) {
- float sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ float sample = dst[n] * factor;
dst[n] = sample / (sqrtf(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
- for (int n = 0; n < nb_samples; n++) {
- float sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ float sample = dst[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
@@ -170,8 +236,8 @@ static void filter_flt(ASoftClipContext *s,
}
break;
case ASC_SIN:
- for (int n = 0; n < nb_samples; n++) {
- float sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ float sample = dst[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
@@ -181,53 +247,86 @@ static void filter_flt(ASoftClipContext *s,
}
break;
case ASC_ERF:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = erff(src[n] * factor);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = erff(dst[n] * factor);
dst[n] *= gain;
}
break;
default:
av_assert0(0);
}
+
+ w = (float *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
+ dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
+
+ for (int n = 0; n < nb_samples; n++)
+ dst[n] = dst[n * oversample] * scale;
}
}
+static inline double run_lowpassd(const Lowpass *const s,
+ double src, double *w)
+{
+ double dst;
+
+ dst = src * s->db0 + w[0];
+ w[0] = s->db1 * src + w[1] - s->da1 * dst;
+ w[1] = s->db2 * src - s->da2 * dst;
+
+ return dst;
+}
+
static void filter_dbl(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
+ const int oversample = s->oversample;
+ const int nb_osamples = nb_samples * oversample;
+ const double scale = oversample > 1 ? oversample * 0.5 : 1.;
double threshold = s->threshold;
double gain = s->output * threshold;
double factor = 1. / threshold;
double param = s->param;
for (int c = start; c < end; c++) {
+ double *w = (double *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
const double *src = sptr[c];
double *dst = dptr[c];
+ for (int n = 0; n < nb_samples; n++) {
+ dst[oversample * n] = src[n];
+
+ for (int m = 1; m < oversample; m++)
+ dst[oversample * n + m] = 0.f;
+ }
+
+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
+ dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
+
switch (s->type) {
case ASC_HARD:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = av_clipd(src[n] * factor, -1., 1.);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = av_clipd(dst[n] * factor, -1., 1.);
dst[n] *= gain;
}
break;
case ASC_TANH:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = tanh(src[n] * factor * param);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = tanh(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = 2. / M_PI * atan(src[n] * factor * param);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
- for (int n = 0; n < nb_samples; n++) {
- double sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ double sample = dst[n] * factor;
if (FFABS(sample) >= 1.5)
dst[n] = FFSIGN(sample);
@@ -237,22 +336,22 @@ static void filter_dbl(ASoftClipContext *s,
}
break;
case ASC_EXP:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
- for (int n = 0; n < nb_samples; n++) {
- double sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ double sample = dst[n] * factor;
dst[n] = sample / (sqrt(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
- for (int n = 0; n < nb_samples; n++) {
- double sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ double sample = dst[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
@@ -262,8 +361,8 @@ static void filter_dbl(ASoftClipContext *s,
}
break;
case ASC_SIN:
- for (int n = 0; n < nb_samples; n++) {
- double sample = src[n] * factor;
+ for (int n = 0; n < nb_osamples; n++) {
+ double sample = dst[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
@@ -273,14 +372,21 @@ static void filter_dbl(ASoftClipContext *s,
}
break;
case ASC_ERF:
- for (int n = 0; n < nb_samples; n++) {
- dst[n] = erf(src[n] * factor);
+ for (int n = 0; n < nb_osamples; n++) {
+ dst[n] = erf(dst[n] * factor);
dst[n] *= gain;
}
break;
default:
av_assert0(0);
}
+
+ w = (double *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
+ dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
+
+ for (int n = 0; n < nb_samples; n++)
+ dst[n] = dst[n * oversample] * scale;
}
}
@@ -288,47 +394,21 @@ static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
- int ret;
switch (inlink->format) {
- case AV_SAMPLE_FMT_FLT:
case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
- case AV_SAMPLE_FMT_DBL:
case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
default: av_assert0(0);
}
- if (s->oversample <= 1)
- return 0;
-
- s->up_ctx = swr_alloc();
- s->down_ctx = swr_alloc();
- if (!s->up_ctx || !s->down_ctx)
+ s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
+ s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
+ if (!s->frame[0] || !s->frame[1])
return AVERROR(ENOMEM);
- av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0);
- av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0);
- av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0);
-
- av_opt_set_int(s->up_ctx, "out_channel_layout", inlink->channel_layout, 0);
- av_opt_set_int(s->up_ctx, "out_sample_rate", inlink->sample_rate * s->oversample, 0);
- av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0);
-
- av_opt_set_int(s->down_ctx, "in_channel_layout", inlink->channel_layout, 0);
- av_opt_set_int(s->down_ctx, "in_sample_rate", inlink->sample_rate * s->oversample, 0);
- av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0);
-
- av_opt_set_int(s->down_ctx, "out_channel_layout", inlink->channel_layout, 0);
- av_opt_set_int(s->down_ctx, "out_sample_rate", inlink->sample_rate, 0);
- av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0);
-
- ret = swr_init(s->up_ctx);
- if (ret < 0)
- return ret;
-
- ret = swr_init(s->down_ctx);
- if (ret < 0)
- return ret;
+ for (int i = 0; i < MAX_OVERSAMPLE; i++) {
+ get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
+ }
return 0;
}
@@ -361,14 +441,14 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
- int ret, nb_samples, channels;
+ int nb_samples, channels;
ThreadData td;
AVFrame *out;
- if (av_frame_is_writable(in)) {
+ if (av_frame_is_writable(in) && s->oversample == 1) {
out = in;
} else {
- out = ff_get_audio_buffer(outlink, in->nb_samples);
+ out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
@@ -376,72 +456,29 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
av_frame_copy_props(out, in);
}
- if (av_sample_fmt_is_planar(in->format)) {
- nb_samples = in->nb_samples;
- channels = in->channels;
- } else {
- nb_samples = in->channels * in->nb_samples;
- channels = 1;
- }
-
- if (s->oversample > 1) {
- s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
- if (!s->frame) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
+ nb_samples = in->nb_samples;
+ channels = in->channels;
- ret = swr_convert(s->up_ctx, (uint8_t**)s->frame->extended_data, in->nb_samples * s->oversample,
- (const uint8_t **)in->extended_data, in->nb_samples);
- if (ret < 0)
- goto fail;
-
- td.in = s->frame;
- td.out = s->frame;
- td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels;
- td.channels = channels;
- ff_filter_execute(ctx, filter_channels, &td, NULL,
- FFMIN(channels, ff_filter_get_nb_threads(ctx)));
-
- ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples,
- (const uint8_t **)s->frame->extended_data, ret);
- if (ret < 0)
- goto fail;
-
- if (out->pts)
- out->pts -= s->delay;
- s->delay += in->nb_samples - ret;
- out->nb_samples = ret;
-
- av_frame_free(&s->frame);
- } else {
- td.in = in;
- td.out = out;
- td.nb_samples = nb_samples;
- td.channels = channels;
- ff_filter_execute(ctx, filter_channels, &td, NULL,
- FFMIN(channels, ff_filter_get_nb_threads(ctx)));
- }
+ td.in = in;
+ td.out = out;
+ td.nb_samples = nb_samples;
+ td.channels = channels;
+ ff_filter_execute(ctx, filter_channels, &td, NULL,
+ FFMIN(channels, ff_filter_get_nb_threads(ctx)));
if (out != in)
av_frame_free(&in);
+ out->nb_samples /= s->oversample;
return ff_filter_frame(outlink, out);
-fail:
- if (out != in)
- av_frame_free(&out);
- av_frame_free(&in);
- av_frame_free(&s->frame);
-
- return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ASoftClipContext *s = ctx->priv;
- swr_free(&s->up_ctx);
- swr_free(&s->down_ctx);
+ av_frame_free(&s->frame[0]);
+ av_frame_free(&s->frame[1]);
}
static const AVFilterPad inputs[] = {
More information about the ffmpeg-cvslog
mailing list