diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
index e21b3e4a04..e05c0a179d 100644
--- a/libavfilter/af_aresample.c
+++ b/libavfilter/af_aresample.c
@@ -230,10 +230,15 @@ static int request_frame(AVFilterLink *outlink)
     if (ret == AVERROR_EOF) {
         AVFrame *outsamplesref;
         int n_out = 4096;
+        int64_t pts;
 
         outsamplesref = ff_get_audio_buffer(outlink, n_out);
         if (!outsamplesref)
             return AVERROR(ENOMEM);
+
+        pts = swr_next_pts(aresample->swr, INT64_MIN);
+        pts = ROUNDED_DIV(pts, inlink->sample_rate);
+
         n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
         if (n_out <= 0) {
             av_frame_free(&outsamplesref);
@@ -242,14 +247,8 @@ static int request_frame(AVFilterLink *outlink)
 
         outsamplesref->sample_rate = outlink->sample_rate;
         outsamplesref->nb_samples  = n_out;
-#if 0
-        outsamplesref->pts = aresample->next_pts;
-        if(aresample->next_pts != AV_NOPTS_VALUE)
-            aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
-#else
-        outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
-        outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
-#endif
+
+        outsamplesref->pts = pts;
 
         return ff_filter_frame(outlink, outsamplesref);
     }