[FFmpeg-devel] [PATCH 1/3] avfilter/af_loudnorm: switch to activate

[FFmpeg-devel] [PATCH 1/3] avfilter/af_loudnorm: switch to activate

[Paul B Mahol]

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavfilter/af_loudnorm.c | 62 ++++++++++++++++++++++++++++++---------
 1 file changed, 48 insertions(+), 14 deletions(-)

diff --git a/libavfilter/af_loudnorm.c b/libavfilter/af_loudnorm.c
index dbe7fba986..5b4b6e8548 100644
--- a/libavfilter/af_loudnorm.c
+++ b/libavfilter/af_loudnorm.c
@@ -22,6 +22,7 @@
 
 #include "libavutil/opt.h"
 #include "avfilter.h"
+#include "filters.h"
 #include "internal.h"
 #include "audio.h"
 #include "ebur128.h"
@@ -502,9 +503,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
         ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length);
 
         s->pts +=
-        out->nb_samples =
-        inlink->min_samples =
-        inlink->max_samples = subframe_length;
+        out->nb_samples = subframe_length;
 
         s->frame_type = INNER_FRAME;
         break;
@@ -636,15 +635,14 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     return ff_filter_frame(outlink, out);
 }
 
-static int request_frame(AVFilterLink *outlink)
+static int flush_frame(AVFilterLink *outlink)
 {
-    int ret;
     AVFilterContext *ctx = outlink->src;
     AVFilterLink *inlink = ctx->inputs[0];
     LoudNormContext *s = ctx->priv;
+    int ret = 0;
 
-    ret = ff_request_frame(inlink);
-    if (ret == AVERROR_EOF && s->frame_type == INNER_FRAME) {
+    if (s->frame_type == INNER_FRAME) {
         double *src;
         double *buf;
         int nb_samples, n, c, offset;
@@ -681,6 +679,48 @@ static int request_frame(AVFilterLink *outlink)
     return ret;
 }
 
+static int activate(AVFilterContext *ctx)
+{
+    AVFilterLink *inlink = ctx->inputs[0];
+    AVFilterLink *outlink = ctx->outputs[0];
+    LoudNormContext *s = ctx->priv;
+    AVFrame *in = NULL;
+    int ret = 0, status;
+    int64_t pts;
+
+    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+    if (s->frame_type != LINEAR_MODE) {
+        int nb_samples;
+
+        if (s->frame_type == FIRST_FRAME) {
+            nb_samples = frame_size(inlink->sample_rate, 3000);
+        } else {
+            nb_samples = frame_size(inlink->sample_rate, 100);
+        }
+
+        ret = ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
+    } else {
+        ret = ff_inlink_consume_frame(inlink, &in);
+    }
+
+    if (ret < 0)
+        return ret;
+    if (ret > 0)
+        ret = filter_frame(inlink, in);
+    if (ret < 0)
+        return ret;
+
+    if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+        ff_outlink_set_status(outlink, status, pts);
+        return flush_frame(outlink);
+    }
+
+    FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+    return FFERROR_NOT_READY;
+}
+
 static int query_formats(AVFilterContext *ctx)
 {
     LoudNormContext *s = ctx->priv;
@@ -749,11 +789,6 @@ static int config_input(AVFilterLink *inlink)
 
     init_gaussian_filter(s);
 
-    if (s->frame_type != LINEAR_MODE) {
-        inlink->min_samples =
-        inlink->max_samples = frame_size(inlink->sample_rate, 3000);
-    }
-
     s->pts = AV_NOPTS_VALUE;
     s->buf_index =
     s->prev_buf_index =
@@ -894,14 +929,12 @@ static const AVFilterPad avfilter_af_loudnorm_inputs[] = {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
-        .filter_frame = filter_frame,
     },
 };
 
 static const AVFilterPad avfilter_af_loudnorm_outputs[] = {
     {
         .name          = "default",
-        .request_frame = request_frame,
         .type          = AVMEDIA_TYPE_AUDIO,
     },
 };
@@ -912,6 +945,7 @@ const AVFilter ff_af_loudnorm = {
     .priv_size     = sizeof(LoudNormContext),
     .priv_class    = &loudnorm_class,
     .init          = init,
+    .activate      = activate,
     .uninit        = uninit,
     FILTER_INPUTS(avfilter_af_loudnorm_inputs),
     FILTER_OUTPUTS(avfilter_af_loudnorm_outputs),
-- 
2.33.0

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[FFmpeg-devel] [PATCH 2/3] avfilter/af_loudnorm: stop rewritting pts

[Paul B Mahol]

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavfilter/af_loudnorm.c | 23 +++++++++++++----------
 1 file changed, 13 insertions(+), 10 deletions(-)

diff --git a/libavfilter/af_loudnorm.c b/libavfilter/af_loudnorm.c
index 5b4b6e8548..7c8ac3a39d 100644
--- a/libavfilter/af_loudnorm.c
+++ b/libavfilter/af_loudnorm.c
@@ -86,7 +86,7 @@ typedef struct LoudNormContext {
     int attack_length;
     int release_length;
 
-    int64_t pts;
+    int64_t pts[30];
     enum FrameType frame_type;
     int above_threshold;
     int prev_nb_samples;
@@ -432,10 +432,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
         av_frame_copy_props(out, in);
     }
 
-    if (s->pts == AV_NOPTS_VALUE)
-        s->pts = in->pts;
+    out->pts = s->pts[0];
+    memmove(s->pts, &s->pts[1], (FF_ARRAY_ELEMS(s->pts) - 1) * sizeof(s->pts[0]));
 
-    out->pts = s->pts;
     src = (const double *)in->data[0];
     dst = (double *)out->data[0];
     buf = s->buf;
@@ -502,7 +501,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
         true_peak_limiter(s, dst, subframe_length, inlink->channels);
         ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length);
 
-        s->pts +=
         out->nb_samples = subframe_length;
 
         s->frame_type = INNER_FRAME;
@@ -567,7 +565,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
         if (s->index >= 30)
             s->index -= 30;
         s->prev_nb_samples = in->nb_samples;
-        s->pts += in->nb_samples;
         break;
 
     case FINAL_FRAME:
@@ -625,13 +622,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 
         dst = (double *)out->data[0];
         ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
-        s->pts += in->nb_samples;
         break;
     }
 
     if (in != out)
         av_frame_free(&in);
-
     return ff_filter_frame(outlink, out);
 }
 
@@ -706,8 +701,17 @@ static int activate(AVFilterContext *ctx)
 
     if (ret < 0)
         return ret;
-    if (ret > 0)
+    if (ret > 0) {
+        if (s->frame_type == FIRST_FRAME) {
+            const int nb_samples = frame_size(inlink->sample_rate, 100);
+
+            for (int i = 0; i < FF_ARRAY_ELEMS(s->pts); i++)
+                s->pts[i] = in->pts + i * nb_samples;
+        } else {
+            s->pts[29] = in->pts;
+        }
         ret = filter_frame(inlink, in);
+    }
     if (ret < 0)
         return ret;
 
@@ -789,7 +793,6 @@ static int config_input(AVFilterLink *inlink)
 
     init_gaussian_filter(s);
 
-    s->pts = AV_NOPTS_VALUE;
     s->buf_index =
     s->prev_buf_index =
     s->limiter_buf_index = 0;
-- 
2.33.0

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[FFmpeg-devel] [PATCH 3/3] avfilter/af_loudnorm: remove hard clipping of samples

[Paul B Mahol]

It can cause unpleasant artifacts.

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavfilter/af_loudnorm.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

diff --git a/libavfilter/af_loudnorm.c b/libavfilter/af_loudnorm.c
index 7c8ac3a39d..9bb0c65bb7 100644
--- a/libavfilter/af_loudnorm.c
+++ b/libavfilter/af_loudnorm.c
@@ -394,12 +394,8 @@ static void true_peak_limiter(LoudNormContext *s, double *out, int nb_samples, i
     } while (smp_cnt < nb_samples);
 
     for (n = 0; n < nb_samples; n++) {
-        for (c = 0; c < channels; c++) {
+        for (c = 0; c < channels; c++)
             out[c] = buf[index + c];
-            if (fabs(out[c]) > ceiling) {
-                out[c] = ceiling * (out[c] < 0 ? -1 : 1);
-            }
-        }
         out += channels;
         index += channels;
         if (index >= s->limiter_buf_size)
-- 
2.33.0

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