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[FFmpeg-devel] avfilter/af_asoftclip: add oversampling support

Message ID 20201105123644.13223-1-onemda@gmail.com
State Accepted
Commit ee4a04654014e24f47f5a31265fb3bde6fad8ce6
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Series [FFmpeg-devel] avfilter/af_asoftclip: add oversampling support
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Commit Message

Paul B Mahol Nov. 5, 2020, 12:36 p.m. UTC
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 configure                  |   1 +
 doc/filters.texi           |   3 ++
 libavfilter/af_asoftclip.c | 106 ++++++++++++++++++++++++++++++++++---
 3 files changed, 103 insertions(+), 7 deletions(-)

Comments

Paul B Mahol Nov. 9, 2020, 10:52 a.m. UTC | #1
Will apply soon.

On Thu, Nov 5, 2020 at 1:36 PM Paul B Mahol <onemda@gmail.com> wrote:

> Signed-off-by: Paul B Mahol <onemda@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.
> +
> +@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
>
>
diff mbox series

Patch

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,