* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#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"
enum ASoftClipTypes {
+ ASC_HARD = -1,
ASC_TANH,
ASC_ATAN,
ASC_CUBIC,
ASC_ALG,
ASC_QUINTIC,
ASC_SIN,
+ ASC_ERF,
NB_TYPES,
};
const AVClass *class;
int type;
+ int oversample;
+ int64_t delay;
+ double threshold;
+ double output;
double param;
+ SwrContext *up_ctx;
+ SwrContext *down_ctx;
+
+ AVFrame *frame;
+
void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
- int nb_samples, int channels);
+ int nb_samples, int channels, int start, int end);
} ASoftClipContext;
#define OFFSET(x) offsetof(ASoftClipContext, x)
-#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_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[] = {
- { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, NB_TYPES-1, A, "types" },
+ { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" },
+ { "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, A, "types" },
{ "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" },
{ "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" },
{ "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" },
{ "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" },
{ "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
{ "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
+ { "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" },
+ { "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 },
{ NULL }
};
return ff_set_common_samplerates(ctx, formats);
}
-#define SQR(x) ((x) * (x))
-
static void filter_flt(ASoftClipContext *s,
void **dptr, const void **sptr,
- int nb_samples, int channels)
+ int nb_samples, int channels,
+ int start, int end)
{
+ float threshold = s->threshold;
+ float gain = s->output * threshold;
+ float factor = 1.f / threshold;
float param = s->param;
- for (int c = 0; c < channels; c++) {
+ for (int c = start; c < end; c++) {
const float *src = sptr[c];
float *dst = dptr[c];
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);
+ dst[n] *= gain;
+ }
+ break;
case ASC_TANH:
for (int n = 0; n < nb_samples; n++) {
- dst[n] = tanhf(src[n] * param);
+ dst[n] = tanhf(src[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] * param);
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
+ dst[n] *= gain;
+ }
break;
case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) {
- if (FFABS(src[n]) >= 1.5f)
- dst[n] = FFSIGN(src[n]);
+ float sample = src[n] * factor;
+
+ if (FFABS(sample) >= 1.5f)
+ dst[n] = FFSIGN(sample);
else
- dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
+ dst[n] = sample - 0.1481f * powf(sample, 3.f);
+ dst[n] *= gain;
}
break;
case ASC_EXP:
- for (int n = 0; n < nb_samples; n++)
- dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
+ dst[n] *= gain;
+ }
break;
case ASC_ALG:
- for (int n = 0; n < nb_samples; n++)
- dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
+ for (int n = 0; n < nb_samples; n++) {
+ float sample = src[n] * factor;
+
+ dst[n] = sample / (sqrtf(param + sample * sample));
+ dst[n] *= gain;
+ }
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) {
- if (FFABS(src[n]) >= 1.25)
- dst[n] = FFSIGN(src[n]);
+ float sample = src[n] * factor;
+
+ if (FFABS(sample) >= 1.25)
+ dst[n] = FFSIGN(sample);
else
- dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
+ dst[n] = sample - 0.08192f * powf(sample, 5.f);
+ dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_samples; n++) {
- if (FFABS(src[n]) >= M_PI_2)
- dst[n] = FFSIGN(src[n]);
+ float sample = src[n] * factor;
+
+ if (FFABS(sample) >= M_PI_2)
+ dst[n] = FFSIGN(sample);
else
- dst[n] = sinf(src[n]);
+ dst[n] = sinf(sample);
+ dst[n] *= gain;
+ }
+ break;
+ case ASC_ERF:
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = erff(src[n] * factor);
+ dst[n] *= gain;
}
break;
+ default:
+ av_assert0(0);
}
}
}
static void filter_dbl(ASoftClipContext *s,
void **dptr, const void **sptr,
- int nb_samples, int channels)
+ int nb_samples, int channels,
+ int start, int end)
{
+ double threshold = s->threshold;
+ double gain = s->output * threshold;
+ double factor = 1. / threshold;
double param = s->param;
- for (int c = 0; c < channels; c++) {
+ for (int c = start; c < end; c++) {
const double *src = sptr[c];
double *dst = dptr[c];
switch (s->type) {
+ case ASC_HARD:
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = av_clipd(src[n] * factor, -1., 1.);
+ dst[n] *= gain;
+ }
+ break;
case ASC_TANH:
for (int n = 0; n < nb_samples; n++) {
- dst[n] = tanh(src[n] * param);
+ dst[n] = tanh(src[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] * param);
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = 2. / M_PI * atan(src[n] * factor * param);
+ dst[n] *= gain;
+ }
break;
case ASC_CUBIC:
for (int n = 0; n < nb_samples; n++) {
- if (FFABS(src[n]) >= 1.5)
- dst[n] = FFSIGN(src[n]);
+ double sample = src[n] * factor;
+
+ if (FFABS(sample) >= 1.5)
+ dst[n] = FFSIGN(sample);
else
- dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
+ dst[n] = sample - 0.1481 * pow(sample, 3.);
+ dst[n] *= gain;
}
break;
case ASC_EXP:
- for (int n = 0; n < nb_samples; n++)
- dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
+ dst[n] *= gain;
+ }
break;
case ASC_ALG:
- for (int n = 0; n < nb_samples; n++)
- dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
+ for (int n = 0; n < nb_samples; n++) {
+ double sample = src[n] * factor;
+
+ dst[n] = sample / (sqrt(param + sample * sample));
+ dst[n] *= gain;
+ }
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_samples; n++) {
- if (FFABS(src[n]) >= 1.25)
- dst[n] = FFSIGN(src[n]);
+ double sample = src[n] * factor;
+
+ if (FFABS(sample) >= 1.25)
+ dst[n] = FFSIGN(sample);
else
- dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
+ dst[n] = sample - 0.08192 * pow(sample, 5.);
+ dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_samples; n++) {
- if (FFABS(src[n]) >= M_PI_2)
- dst[n] = FFSIGN(src[n]);
+ double sample = src[n] * factor;
+
+ if (FFABS(sample) >= M_PI_2)
+ dst[n] = FFSIGN(sample);
else
- dst[n] = sin(src[n]);
+ dst[n] = sin(sample);
+ dst[n] *= gain;
}
break;
+ case ASC_ERF:
+ for (int n = 0; n < nb_samples; n++) {
+ dst[n] = erf(src[n] * factor);
+ dst[n] *= gain;
+ }
+ break;
+ default:
+ av_assert0(0);
}
}
}
{
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)
+ 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;
+
+ return 0;
+}
+
+typedef struct ThreadData {
+ AVFrame *in, *out;
+ int nb_samples;
+ int channels;
+} ThreadData;
+
+static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ ASoftClipContext *s = ctx->priv;
+ ThreadData *td = arg;
+ AVFrame *out = td->out;
+ AVFrame *in = td->in;
+ const int channels = td->channels;
+ const int nb_samples = td->nb_samples;
+ const int start = (channels * jobnr) / nb_jobs;
+ const int end = (channels * (jobnr+1)) / nb_jobs;
+
+ s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
+ nb_samples, channels, start, end);
+
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
- AVFilterLink *outlink = ctx->outputs[0];
ASoftClipContext *s = ctx->priv;
- int nb_samples, channels;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int ret, nb_samples, channels;
+ ThreadData td;
AVFrame *out;
if (av_frame_is_writable(in)) {
channels = 1;
}
- s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
- nb_samples, channels);
+ if (s->oversample > 1) {
+ s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
+ if (!s->frame) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ 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;
+ ctx->internal->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;
+ ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels,
+ ff_filter_get_nb_threads(ctx)));
+ }
if (out != in)
av_frame_free(&in);
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);
}
static const AVFilterPad inputs[] = {
{ NULL }
};
-AVFilter ff_af_asoftclip = {
+const AVFilter ff_af_asoftclip = {
.name = "asoftclip",
.description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
.query_formats = query_formats,
.priv_class = &asoftclip_class,
.inputs = inputs,
.outputs = outputs,
- .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
+ .uninit = uninit,
+ .process_command = ff_filter_process_command,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
+ AVFILTER_FLAG_SLICE_THREADS,
};