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authorPaul B Mahol <onemda@gmail.com>2020-11-05 13:33:32 +0100
committerPaul B Mahol <onemda@gmail.com>2020-11-10 15:25:18 +0100
commitee4a04654014e24f47f5a31265fb3bde6fad8ce6 (patch)
tree231f1ea2a94cf6370cf307a6b2e00ed0e82787b6
parent7e8306dd2da54fa499ea79a74f0ba81507e6c9d4 (diff)
downloadffmpeg-ee4a04654014e24f47f5a31265fb3bde6fad8ce6.tar.gz
avfilter/af_asoftclip: add oversampling support
-rwxr-xr-xconfigure1
-rw-r--r--doc/filters.texi3
-rw-r--r--libavfilter/af_asoftclip.c106
3 files changed, 103 insertions, 7 deletions
diff --git a/configure b/configure
index 8a9e9b3cd7..2f02d7f5c8 100755
--- a/configure
+++ b/configure
@@ -3501,6 +3501,7 @@ afir_filter_deps="avcodec"
afir_filter_select="rdft"
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/doc/filters.texi b/doc/filters.texi
index 40f8f614fe..8380f6cac2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2356,6 +2356,9 @@ It accepts the following values:
@item param
Set additional parameter which controls sigmoid function.
+
+@item oversample
+Set oversampling factor.
@end table
@subsection Commands
diff --git a/libavfilter/af_asoftclip.c b/libavfilter/af_asoftclip.c
index ce1f7ea96a..aaae3c6d4b 100644
--- a/libavfilter/af_asoftclip.c
+++ b/libavfilter/af_asoftclip.c
@@ -21,6 +21,7 @@
#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"
@@ -42,14 +43,22 @@ typedef struct ASoftClipContext {
const AVClass *class;
int type;
+ int oversample;
+ int64_t delay;
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 start, int end);
} 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" },
@@ -63,6 +72,7 @@ static const AVOption asoftclip_options[] = {
{ "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" },
{ "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 }
};
@@ -242,6 +252,7 @@ static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
+ int ret;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLT:
@@ -251,6 +262,38 @@ static int config_input(AVFilterLink *inlink)
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;
}
@@ -280,8 +323,9 @@ static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jo
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
+ ASoftClipContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
- int nb_samples, channels;
+ int ret, nb_samples, channels;
ThreadData td;
AVFrame *out;
@@ -304,17 +348,64 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
channels = 1;
}
- 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 (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[] = {
@@ -343,6 +434,7 @@ AVFilter ff_af_asoftclip = {
.priv_class = &asoftclip_class,
.inputs = inputs,
.outputs = outputs,
+ .uninit = uninit,
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,