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avfilter/af_asoftclip: add two more useful options for finer filtering
This commit is contained in:
Paul B Mahol
parent
1eb751955e
commit
7e3f20c43c
@ -2454,6 +2454,12 @@ It accepts the following values:
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@item erf
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@end table
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@item threshold
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Set threshold from where to start clipping. Default value is 0dB or 1.
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@item output
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Set gain applied to output. Default value is 0dB or 1.
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@item param
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Set additional parameter which controls sigmoid function.
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@ -45,6 +45,8 @@ typedef struct ASoftClipContext {
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int type;
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int oversample;
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int64_t delay;
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double threshold;
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double output;
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double param;
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SwrContext *up_ctx;
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@ -71,6 +73,8 @@ static const AVOption asoftclip_options[] = {
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{ "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
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{ "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
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{ "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" },
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{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
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{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
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{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
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{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
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{ NULL }
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@ -108,13 +112,14 @@ static int query_formats(AVFilterContext *ctx)
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return ff_set_common_samplerates(ctx, formats);
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}
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#define SQR(x) ((x) * (x))
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static void filter_flt(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels,
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int start, int end)
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{
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float threshold = s->threshold;
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float gain = s->output * threshold;
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float factor = 1.f / threshold;
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float param = s->param;
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for (int c = start; c < end; c++) {
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@ -124,53 +129,73 @@ static void filter_flt(ASoftClipContext *s,
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switch (s->type) {
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case ASC_HARD:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = av_clipf(src[n], -1.f, 1.f);
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dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
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dst[n] *= gain;
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}
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break;
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case ASC_TANH:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = tanhf(src[n] * param);
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dst[n] = tanhf(src[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2.f / M_PI * atanf(src[n] * param);
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.5f)
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dst[n] = FFSIGN(src[n]);
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float sample = src[n] * factor;
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if (FFABS(sample) >= 1.5f)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
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dst[n] = sample - 0.1481f * powf(sample, 3.f);
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dst[n] *= gain;
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
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dst[n] *= gain;
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}
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
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for (int n = 0; n < nb_samples; n++) {
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float sample = src[n] * factor;
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dst[n] = sample / (sqrtf(param + sample * sample));
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dst[n] *= gain;
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}
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.25)
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dst[n] = FFSIGN(src[n]);
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float sample = src[n] * factor;
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if (FFABS(sample) >= 1.25)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
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dst[n] = sample - 0.08192f * powf(sample, 5.f);
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dst[n] *= gain;
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= M_PI_2)
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dst[n] = FFSIGN(src[n]);
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float sample = src[n] * factor;
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if (FFABS(sample) >= M_PI_2)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sinf(src[n]);
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dst[n] = sinf(sample);
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dst[n] *= gain;
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}
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break;
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case ASC_ERF:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = erff(src[n]);
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dst[n] = erff(src[n] * factor);
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dst[n] *= gain;
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}
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break;
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default:
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@ -184,6 +209,9 @@ static void filter_dbl(ASoftClipContext *s,
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int nb_samples, int channels,
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int start, int end)
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{
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double threshold = s->threshold;
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double gain = s->output * threshold;
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double factor = 1. / threshold;
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double param = s->param;
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for (int c = start; c < end; c++) {
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@ -193,53 +221,73 @@ static void filter_dbl(ASoftClipContext *s,
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switch (s->type) {
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case ASC_HARD:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = av_clipd(src[n], -1., 1.);
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dst[n] = av_clipd(src[n] * factor, -1., 1.);
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dst[n] *= gain;
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}
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break;
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case ASC_TANH:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = tanh(src[n] * param);
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dst[n] = tanh(src[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2. / M_PI * atan(src[n] * param);
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2. / M_PI * atan(src[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.5)
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dst[n] = FFSIGN(src[n]);
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double sample = src[n] * factor;
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if (FFABS(sample) >= 1.5)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
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dst[n] = sample - 0.1481 * pow(sample, 3.);
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dst[n] *= gain;
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
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dst[n] *= gain;
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}
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_samples; n++)
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dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
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for (int n = 0; n < nb_samples; n++) {
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double sample = src[n] * factor;
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dst[n] = sample / (sqrt(param + sample * sample));
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dst[n] *= gain;
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}
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= 1.25)
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dst[n] = FFSIGN(src[n]);
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double sample = src[n] * factor;
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if (FFABS(sample) >= 1.25)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
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dst[n] = sample - 0.08192 * pow(sample, 5.);
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dst[n] *= gain;
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_samples; n++) {
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if (FFABS(src[n]) >= M_PI_2)
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dst[n] = FFSIGN(src[n]);
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double sample = src[n] * factor;
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if (FFABS(sample) >= M_PI_2)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sin(src[n]);
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dst[n] = sin(sample);
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dst[n] *= gain;
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}
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break;
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case ASC_ERF:
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = erf(src[n]);
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dst[n] = erf(src[n] * factor);
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dst[n] *= gain;
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}
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break;
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default:
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