avfilter/af_astats: add slice threading support

[ffmpeg] / libavfilter / avf_ahistogram.c

diff --git a/libavfilter/avf_ahistogram.c b/libavfilter/avf_ahistogram.c
index 587415175b562d7e87389bf4dca151245495db1d..92cda46756144976658bd5dd5817956e069593dc 100644 (file)
--- a/libavfilter/avf_ahistogram.c
+++ b/libavfilter/avf_ahistogram.c
@@ -22,6 +22,7 @@
 #include "libavutil/opt.h"
 #include "libavutil/parseutils.h"
 #include "avfilter.h"
+#include "filters.h"
 #include "formats.h"
 #include "audio.h"
 #include "video.h"
@@ -54,6 +55,7 @@ typedef struct AudioHistogramContext {
     float *combine_buffer;
     AVFrame *in[101];
     int first;
+    int nb_samples;
 } AudioHistogramContext;
 
 #define OFFSET(x) offsetof(AudioHistogramContext, x)
@@ -117,13 +119,8 @@ static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioHistogramContext *s = ctx->priv;
-    int nb_samples;
-
-    nb_samples = FFMAX(1024, ((double)inlink->sample_rate / av_q2d(s->frame_rate)) + 0.5);
-    inlink->partial_buf_size =
-    inlink->min_samples =
-    inlink->max_samples = nb_samples;
 
+    s->nb_samples = FFMAX(1, av_rescale(inlink->sample_rate, s->frame_rate.den, s->frame_rate.num));
     s->dchannels = s->dmode == SINGLE ? 1 : inlink->channels;
     s->shistogram = av_calloc(s->w, s->dchannels * sizeof(*s->shistogram));
     if (!s->shistogram)
@@ -166,6 +163,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     const int w = s->w;
     int c, y, n, p, bin;
     uint64_t acmax = 1;
+    AVFrame *clone;
 
     if (!s->out || s->out->width  != outlink->w ||
                    s->out->height != outlink->h) {
@@ -366,7 +364,33 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
             s->ypos = H;
     }
 
-    return ff_filter_frame(outlink, av_frame_clone(s->out));
+    clone = av_frame_clone(s->out);
+    if (!clone)
+        return AVERROR(ENOMEM);
+
+    return ff_filter_frame(outlink, clone);
+}
+
+static int activate(AVFilterContext *ctx)
+{
+    AVFilterLink *inlink = ctx->inputs[0];
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioHistogramContext *s = ctx->priv;
+    AVFrame *in;
+    int ret;
+
+    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+    ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
+    if (ret < 0)
+        return ret;
+    if (ret > 0)
+        return filter_frame(inlink, in);
+
+    FF_FILTER_FORWARD_STATUS(inlink, outlink);
+    FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+    return FFERROR_NOT_READY;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
@@ -382,17 +406,16 @@ static av_cold void uninit(AVFilterContext *ctx)
         av_frame_free(&s->in[i]);
 }
 
-static const AVFilterPad audiovectorscope_inputs[] = {
+static const AVFilterPad ahistogram_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
-        .filter_frame = filter_frame,
     },
     { NULL }
 };
 
-static const AVFilterPad audiovectorscope_outputs[] = {
+static const AVFilterPad ahistogram_outputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_VIDEO,
@@ -407,7 +430,8 @@ AVFilter ff_avf_ahistogram = {
     .uninit        = uninit,
     .query_formats = query_formats,
     .priv_size     = sizeof(AudioHistogramContext),
-    .inputs        = audiovectorscope_inputs,
-    .outputs       = audiovectorscope_outputs,
+    .activate      = activate,
+    .inputs        = ahistogram_inputs,
+    .outputs       = ahistogram_outputs,
     .priv_class    = &ahistogram_class,
 };