diff --git a/audio/out/push.c b/audio/out/push.c
index 225b8b22d2..8ec8ba28ca 100644
--- a/audio/out/push.c
+++ b/audio/out/push.c
@@ -49,6 +49,7 @@ struct ao_push_state {
 
     struct mp_audio_buffer *buffer;
 
+    bool newdata;
     bool terminate;
     bool playing;
 
@@ -185,6 +186,8 @@ static int play(struct ao *ao, void **data, int samples, int flags)
         audio.planes[n] = data[n];
     audio.samples = write_samples;
     mp_audio_buffer_append(p->buffer, &audio);
+    if (write_samples > 0)
+        p->newdata = true;
 
     p->final_chunk = !!(flags & AOPLAY_FINAL_CHUNK);
     p->playing = true;
@@ -216,8 +219,10 @@ static int ao_play_data(struct ao *ao)
         MP_WARN(ao, "Audio device returned non-sense value.\n");
         r = data.samples;
     }
-    if (r > 0)
+    if (r > 0) {
         mp_audio_buffer_skip(p->buffer, r);
+        p->newdata = false;
+    }
     if (p->final_chunk && mp_audio_buffer_samples(p->buffer) == 0) {
         p->playing = false;
         p->expected_end_time = mp_time_sec() + AO_EOF_DELAY + 0.25; // + margin
@@ -256,7 +261,7 @@ static void *playthread(void *arg)
             }
             // Half of the buffer played -> wakeup playback thread to get more.
             double min_wait = ao->device_buffer / (double)ao->samplerate;
-            if (timeout <= min_wait / 2 + 0.001)
+            if (timeout <= min_wait / 2 + 0.001 && !p->newdata)
                 mp_input_wakeup(ao->input_ctx);
             // Avoid wasting CPU - this assumes ao_play_data() usually fills the
             // audio buffer as far as possible, so even if the device buffer