diff --git a/audio/out/ao.c b/audio/out/ao.c
index 32ffc3d82d..52a38b63be 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -204,8 +204,6 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
 
     init_buffer_pre(ao);
 
-    ao->period_size = 1;
-
     int r = ao->driver->init(ao);
     if (r < 0) {
         // Silly exception for coreaudio spdif redirection
@@ -222,11 +220,6 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
     }
     ao->driver_initialized = true;
 
-    if (ao->period_size < 1) {
-        MP_ERR(ao, "Invalid period size set.\n");
-        goto fail;
-    }
-
     ao->sstride = af_fmt_to_bytes(ao->format);
     ao->num_planes = 1;
     if (af_fmt_is_planar(ao->format)) {
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 0380162eb4..49bdd32835 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -850,7 +850,6 @@ static int init_device(struct ao *ao, int mode)
     MP_VERBOSE(ao, "period size: %d samples\n", (int)p->outburst);
 
     ao->device_buffer = p->buffersize;
-    ao->period_size = p->outburst;
 
     p->convert.channels = ao->channels.num;
 
@@ -998,6 +997,8 @@ static bool recover_and_get_state(struct ao *ao, struct mp_pcm_state *state)
         state->delay = MPMAX(del, 0) / (double)ao->samplerate;
         state->free_samples = snd_pcm_status_get_avail(st);
         state->free_samples = MPCLAMP(state->free_samples, 0, ao->device_buffer);
+        // Align to period size.
+        state->free_samples = state->free_samples / p->outburst * p->outburst;
         state->queued_samples = ao->device_buffer - state->free_samples;
         state->playing = pcmst == SND_PCM_STATE_RUNNING ||
                          pcmst == SND_PCM_STATE_PAUSED;
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 049f8df2cf..8a98f0b67a 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -158,7 +158,6 @@ static int init(struct ao *ao)
     ao->untimed = true;
 
     ao->device_buffer = ac->aframesize * ac->framecount;
-    ao->period_size = ao->device_buffer;
 
     ac->filter_root = mp_filter_create_root(ao->global);
     ac->fix_frame_size = mp_fixed_aframe_size_create(ac->filter_root,
diff --git a/audio/out/ao_null.c b/audio/out/ao_null.c
index 2f58ea22c3..c8c4117ec0 100644
--- a/audio/out/ao_null.c
+++ b/audio/out/ao_null.c
@@ -109,8 +109,6 @@ static int init(struct ao *ao)
 
     priv->last_time = mp_time_sec();
 
-    ao->period_size = priv->outburst;
-
     return 0;
 }
 
diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
index 64d957e893..0c5ec2d7f1 100644
--- a/audio/out/ao_openal.c
+++ b/audio/out/ao_openal.c
@@ -267,7 +267,6 @@ static int init(struct ao *ao)
         goto err_out;
     }
 
-    ao->period_size = p->num_samples;
     return 0;
 
 err_out:
diff --git a/audio/out/buffer.c b/audio/out/buffer.c
index 4c08a26b0b..9cd7717000 100644
--- a/audio/out/buffer.c
+++ b/audio/out/buffer.c
@@ -619,7 +619,7 @@ static bool ao_play_data(struct ao *ao)
     if (got_eof)
         goto eof;
 
-    return samples > 0;
+    return samples > 0 && (samples < space || ao->untimed);
 
 eof:
     MP_VERBOSE(ao, "audio end or underrun\n");
@@ -642,14 +642,14 @@ static void *playthread(void *arg)
     while (1) {
         pthread_mutex_lock(&p->lock);
 
-        bool progress = false;
+        bool retry = false;
         if (!ao->driver->initially_blocked || p->initial_unblocked)
-            progress = ao_play_data(ao);
+            retry = ao_play_data(ao);
 
         // Wait until the device wants us to write more data to it.
         // Fallback to guessing.
         double timeout = INFINITY;
-        if (p->streaming && !p->paused && !progress) {
+        if (p->streaming && !p->paused && !retry) {
             // Wake up again if half of the audio buffer has been played.
             // Since audio could play at a faster or slower pace, wake up twice
             // as often as ideally needed.
@@ -663,7 +663,7 @@ static void *playthread(void *arg)
             pthread_mutex_unlock(&p->pt_lock);
             break;
         }
-        if (!p->need_wakeup && !progress) {
+        if (!p->need_wakeup && !retry) {
             MP_STATS(ao, "start audio wait");
             struct timespec ts = mp_rel_time_to_timespec(timeout);
             pthread_cond_timedwait(&p->pt_wakeup, &p->pt_lock, &ts);
diff --git a/audio/out/internal.h b/audio/out/internal.h
index 5b922657d0..6eb7016f2d 100644
--- a/audio/out/internal.h
+++ b/audio/out/internal.h
@@ -48,12 +48,6 @@ struct ao {
     int init_flags; // AO_INIT_* flags
     bool stream_silence;        // if audio inactive, just play silence
 
-    // Set by the driver on init.
-    // This value is in complete samples (i.e. 1 for stereo means 1 sample
-    // for both channels each).
-    // Used for push based API only.
-    int period_size;
-
     // The device as selected by the user, usually using ao_device_desc.name
     // from an entry from the list returned by driver->list_devices. If the
     // default device should be used, this is set to NULL.
@@ -83,7 +77,12 @@ bool init_buffer_post(struct ao *ao);
 struct mp_pcm_state {
     // Note: free_samples+queued_samples <= ao->device_buffer; the sum may be
     //       less if the audio API can report partial periods played, while
-    //       free_samples should be period-size aligned.
+    //       free_samples should be period-size aligned. If free_samples is not
+    //       period-size aligned, the AO thread might get into a situation where
+    //       it writes a very small number of samples in each iteration, leading
+    //       to extremely inefficient behavior.
+    //       Keep in mind that write() may write less than free_samples (or your
+    //       period size alignment) anyway.
     int free_samples;       // number of free space in ring buffer
     int queued_samples;     // number of samples to play in ring buffer
     double delay;           // total latency in seconds (includes queued_samples)