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avfilter/af_asoftclip: rewrite oversampling

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Fixes most aliasing issues.
This commit is contained in:
Paul B Mahol
2021-09-12 00:31:13 +02:00
parent 3e127b595a
commit 94d4cc24c3
2 changed files with 171 additions and 135 deletions
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1
configure vendored
View File

@@ -3548,7 +3548,6 @@ afir_filter_select="rdft"
ametadata_filter_deps="avformat" ametadata_filter_deps="avformat"
amovie_filter_deps="avcodec avformat" amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample" aresample_filter_deps="swresample"
asoftclip_filter_deps="swresample"
asr_filter_deps="pocketsphinx" asr_filter_deps="pocketsphinx"
ass_filter_deps="libass" ass_filter_deps="libass"
atempo_filter_deps="avcodec" atempo_filter_deps="avcodec"

305
libavfilter/af_asoftclip.c
View File

@@ -21,11 +21,12 @@
#include "libavutil/avassert.h" #include "libavutil/avassert.h"
#include "libavutil/channel_layout.h" #include "libavutil/channel_layout.h"
#include "libavutil/opt.h" #include "libavutil/opt.h"
#include "libswresample/swresample.h"
#include "avfilter.h" #include "avfilter.h"
#include "audio.h" #include "audio.h"
#include "formats.h" #include "formats.h"
#define MAX_OVERSAMPLE 64
enum ASoftClipTypes { enum ASoftClipTypes {
ASC_HARD = -1, ASC_HARD = -1,
ASC_TANH, ASC_TANH,
@@ -39,6 +40,14 @@ enum ASoftClipTypes {
NB_TYPES, 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 { typedef struct ASoftClipContext {
const AVClass *class; const AVClass *class;
@@ -49,10 +58,8 @@ typedef struct ASoftClipContext {
double output; double output;
double param; double param;
SwrContext *up_ctx; Lowpass lowpass[MAX_OVERSAMPLE];
SwrContext *down_ctx; AVFrame *frame[2];
AVFrame *frame;
void (*filter)(struct ASoftClipContext *s, void **dst, const void **src, void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
int nb_samples, int channels, int start, int end); int nb_samples, int channels, int start, int end);
@@ -60,7 +67,6 @@ typedef struct ASoftClipContext {
#define OFFSET(x) offsetof(ASoftClipContext, x) #define OFFSET(x) offsetof(ASoftClipContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM #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[] = { static const AVOption asoftclip_options[] = {
{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" }, { "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 }, { "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 }, { "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 }, { "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 } { NULL }
}; };
@@ -85,8 +91,7 @@ AVFILTER_DEFINE_CLASS(asoftclip);
static int query_formats(AVFilterContext *ctx) static int query_formats(AVFilterContext *ctx)
{ {
static const enum AVSampleFormat sample_fmts[] = { static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE AV_SAMPLE_FMT_NONE
}; };
int ret = ff_set_common_formats_from_list(ctx, sample_fmts); 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); 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, static void filter_flt(ASoftClipContext *s,
void **dptr, const void **sptr, void **dptr, const void **sptr,
int nb_samples, int channels, int nb_samples, int channels,
int start, int end) 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 threshold = s->threshold;
float gain = s->output * threshold; float gain = s->output * threshold;
float factor = 1.f / threshold; float factor = 1.f / threshold;
float param = s->param; float param = s->param;
for (int c = start; c < end; c++) { for (int c = start; c < end; c++) {
float *w = (float *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
const float *src = sptr[c]; const float *src = sptr[c];
float *dst = dptr[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) { switch (s->type) {
case ASC_HARD: case ASC_HARD:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = av_clipf(src[n] * factor, -1.f, 1.f); dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_TANH: case ASC_TANH:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = tanhf(src[n] * factor * param); dst[n] = tanhf(dst[n] * factor * param);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_ATAN: case ASC_ATAN:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2.f / M_PI * atanf(src[n] * factor * param); dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_CUBIC: case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
float sample = src[n] * factor; float sample = dst[n] * factor;
if (FFABS(sample) >= 1.5f) if (FFABS(sample) >= 1.5f)
dst[n] = FFSIGN(sample); dst[n] = FFSIGN(sample);
@@ -145,22 +211,22 @@ static void filter_flt(ASoftClipContext *s,
} }
break; break;
case ASC_EXP: case ASC_EXP:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.; dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_ALG: case ASC_ALG:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
float sample = src[n] * factor; float sample = dst[n] * factor;
dst[n] = sample / (sqrtf(param + sample * sample)); dst[n] = sample / (sqrtf(param + sample * sample));
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_QUINTIC: case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
float sample = src[n] * factor; float sample = dst[n] * factor;
if (FFABS(sample) >= 1.25) if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample); dst[n] = FFSIGN(sample);
@@ -170,8 +236,8 @@ static void filter_flt(ASoftClipContext *s,
} }
break; break;
case ASC_SIN: case ASC_SIN:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
float sample = src[n] * factor; float sample = dst[n] * factor;
if (FFABS(sample) >= M_PI_2) if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample); dst[n] = FFSIGN(sample);
@@ -181,53 +247,86 @@ static void filter_flt(ASoftClipContext *s,
} }
break; break;
case ASC_ERF: case ASC_ERF:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = erff(src[n] * factor); dst[n] = erff(dst[n] * factor);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
default: default:
av_assert0(0); 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, static void filter_dbl(ASoftClipContext *s,
void **dptr, const void **sptr, void **dptr, const void **sptr,
int nb_samples, int channels, int nb_samples, int channels,
int start, int end) 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 threshold = s->threshold;
double gain = s->output * threshold; double gain = s->output * threshold;
double factor = 1. / threshold; double factor = 1. / threshold;
double param = s->param; double param = s->param;
for (int c = start; c < end; c++) { for (int c = start; c < end; c++) {
double *w = (double *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
const double *src = sptr[c]; const double *src = sptr[c];
double *dst = dptr[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) { switch (s->type) {
case ASC_HARD: case ASC_HARD:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = av_clipd(src[n] * factor, -1., 1.); dst[n] = av_clipd(dst[n] * factor, -1., 1.);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_TANH: case ASC_TANH:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = tanh(src[n] * factor * param); dst[n] = tanh(dst[n] * factor * param);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_ATAN: case ASC_ATAN:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2. / M_PI * atan(src[n] * factor * param); dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_CUBIC: case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
double sample = src[n] * factor; double sample = dst[n] * factor;
if (FFABS(sample) >= 1.5) if (FFABS(sample) >= 1.5)
dst[n] = FFSIGN(sample); dst[n] = FFSIGN(sample);
@@ -237,22 +336,22 @@ static void filter_dbl(ASoftClipContext *s,
} }
break; break;
case ASC_EXP: case ASC_EXP:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.; dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_ALG: case ASC_ALG:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
double sample = src[n] * factor; double sample = dst[n] * factor;
dst[n] = sample / (sqrt(param + sample * sample)); dst[n] = sample / (sqrt(param + sample * sample));
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
case ASC_QUINTIC: case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
double sample = src[n] * factor; double sample = dst[n] * factor;
if (FFABS(sample) >= 1.25) if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample); dst[n] = FFSIGN(sample);
@@ -262,8 +361,8 @@ static void filter_dbl(ASoftClipContext *s,
} }
break; break;
case ASC_SIN: case ASC_SIN:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
double sample = src[n] * factor; double sample = dst[n] * factor;
if (FFABS(sample) >= M_PI_2) if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample); dst[n] = FFSIGN(sample);
@@ -273,14 +372,21 @@ static void filter_dbl(ASoftClipContext *s,
} }
break; break;
case ASC_ERF: case ASC_ERF:
for (int n = 0; n < nb_samples; n++) { for (int n = 0; n < nb_osamples; n++) {
dst[n] = erf(src[n] * factor); dst[n] = erf(dst[n] * factor);
dst[n] *= gain; dst[n] *= gain;
} }
break; break;
default: default:
av_assert0(0); 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; AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv; ASoftClipContext *s = ctx->priv;
int ret;
switch (inlink->format) { switch (inlink->format) {
case AV_SAMPLE_FMT_FLT:
case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break; 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; case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
default: av_assert0(0); default: av_assert0(0);
} }
if (s->oversample <= 1) s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
return 0; s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
if (!s->frame[0] || !s->frame[1])
s->up_ctx = swr_alloc();
s->down_ctx = swr_alloc();
if (!s->up_ctx || !s->down_ctx)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0); for (int i = 0; i < MAX_OVERSAMPLE; i++) {
av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0); get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
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;
return 0; return 0;
} }
@@ -361,14 +441,14 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFilterContext *ctx = inlink->dst; AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv; ASoftClipContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0]; AVFilterLink *outlink = ctx->outputs[0];
int ret, nb_samples, channels; int nb_samples, channels;
ThreadData td; ThreadData td;
AVFrame *out; AVFrame *out;
if (av_frame_is_writable(in)) { if (av_frame_is_writable(in) && s->oversample == 1) {
out = in; out = in;
} else { } else {
out = ff_get_audio_buffer(outlink, in->nb_samples); out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
if (!out) { if (!out) {
av_frame_free(&in); av_frame_free(&in);
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
@@ -376,72 +456,29 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
av_frame_copy_props(out, in); av_frame_copy_props(out, in);
} }
if (av_sample_fmt_is_planar(in->format)) { nb_samples = in->nb_samples;
nb_samples = in->nb_samples; channels = in->channels;
channels = in->channels;
} else {
nb_samples = in->channels * in->nb_samples;
channels = 1;
}
if (s->oversample > 1) { td.in = in;
s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample); td.out = out;
if (!s->frame) { td.nb_samples = nb_samples;
ret = AVERROR(ENOMEM); td.channels = channels;
goto fail; ff_filter_execute(ctx, filter_channels, &td, NULL,
} FFMIN(channels, ff_filter_get_nb_threads(ctx)));
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)));
}
if (out != in) if (out != in)
av_frame_free(&in); av_frame_free(&in);
out->nb_samples /= s->oversample;
return ff_filter_frame(outlink, out); 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) static av_cold void uninit(AVFilterContext *ctx)
{ {
ASoftClipContext *s = ctx->priv; ASoftClipContext *s = ctx->priv;
swr_free(&s->up_ctx); av_frame_free(&s->frame[0]);
swr_free(&s->down_ctx); av_frame_free(&s->frame[1]);
} }
static const AVFilterPad inputs[] = { static const AVFilterPad inputs[] = {