[FFmpeg-devel] [PATCH] avfilter/af_asoftclip: add oversampling support
Paul B Mahol
onemda at gmail.com
Mon Nov 9 12:52:18 EET 2020
Will apply soon.
On Thu, Nov 5, 2020 at 1:36 PM Paul B Mahol <onemda at gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> configure | 1 +
> doc/filters.texi | 3 ++
> libavfilter/af_asoftclip.c | 106 ++++++++++++++++++++++++++++++++++---
> 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.
> +
> + at 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,
> --
> 2.17.1
>
>
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