diff --git a/audio/out/ao.c b/audio/out/ao.c
index 480dad69e0..7b301cd2e7 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -200,13 +200,10 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
                af_fmt_to_str(ao->format));
 
     ao->device = talloc_strdup(ao, dev);
-
-    ao->api = ao->driver->play ? &ao_api_push : &ao_api_pull;
-    ao->api_priv = talloc_zero_size(ao, ao->api->priv_size);
-    assert(!ao->api->priv_defaults && !ao->api->options);
-
     ao->stream_silence = flags & AO_INIT_STREAM_SILENCE;
 
+    init_buffer_pre(ao);
+
     ao->period_size = 1;
 
     int r = ao->driver->init(ao);
@@ -216,13 +213,14 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
             char redirect[80], rdevice[80];
             snprintf(redirect, sizeof(redirect), "%s", ao->redirect);
             snprintf(rdevice, sizeof(rdevice), "%s", ao->device ? ao->device : "");
-            talloc_free(ao);
+            ao_uninit(ao);
             return ao_init(probing, global, wakeup_cb, wakeup_ctx,
                            encode_lavc_ctx, flags, samplerate, format, channels,
                            rdevice, redirect);
         }
         goto fail;
     }
+    ao->driver_initialized = true;
 
     if (ao->period_size < 1) {
         MP_ERR(ao, "Invalid period size set.\n");
@@ -249,12 +247,12 @@ static struct ao *ao_init(bool probing, struct mpv_global *global,
     ao->buffer = (ao->buffer + align - 1) / align * align;
     MP_VERBOSE(ao, "using soft-buffer of %d samples.\n", ao->buffer);
 
-    if (ao->api->init(ao) < 0)
+    if (!init_buffer_post(ao))
         goto fail;
     return ao;
 
 fail:
-    talloc_free(ao);
+    ao_uninit(ao);
     return NULL;
 }
 
@@ -348,86 +346,6 @@ struct ao *ao_init_best(struct mpv_global *global,
     return ao;
 }
 
-// Uninitialize and destroy the AO. Remaining audio must be dropped.
-void ao_uninit(struct ao *ao)
-{
-    if (ao)
-        ao->api->uninit(ao);
-    talloc_free(ao);
-}
-
-// Queue the given audio data. Start playback if it hasn't started yet. Return
-// the number of samples that was accepted (the core will try to queue the rest
-// again later). Should never block.
-//  data: start pointer for each plane. If the audio data is packed, only
-//        data[0] is valid, otherwise there is a plane for each channel.
-//  samples: size of the audio data (see ao->sstride)
-//  flags: currently AOPLAY_FINAL_CHUNK can be set
-int ao_play(struct ao *ao, void **data, int samples, int flags)
-{
-    return ao->api->play(ao, data, samples, flags);
-}
-
-int ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
-{
-    return ao->api->control ? ao->api->control(ao, cmd, arg) : CONTROL_UNKNOWN;
-}
-
-// Return size of the buffered data in seconds. Can include the device latency.
-// Basically, this returns how much data there is still to play, and how long
-// it takes until the last sample in the buffer reaches the speakers. This is
-// used for audio/video synchronization, so it's very important to implement
-// this correctly.
-double ao_get_delay(struct ao *ao)
-{
-    return ao->api->get_delay(ao);
-}
-
-// Return free size of the internal audio buffer. This controls how much audio
-// the core should decode and try to queue with ao_play().
-int ao_get_space(struct ao *ao)
-{
-    return ao->api->get_space(ao);
-}
-
-// Stop playback and empty buffers. Essentially go back to the state after
-// ao->init().
-void ao_reset(struct ao *ao)
-{
-    if (ao->api->reset)
-        ao->api->reset(ao);
-    atomic_fetch_and(&ao->events_, ~(unsigned int)AO_EVENT_UNDERRUN);
-}
-
-// Pause playback. Keep the current buffer. ao_get_delay() must return the
-// same value as before pausing.
-void ao_pause(struct ao *ao)
-{
-    if (ao->api->pause)
-        ao->api->pause(ao);
-}
-
-// Resume playback. Play the remaining buffer. If the driver doesn't support
-// pausing, it has to work around this and e.g. use ao_play_silence() to fill
-// the lost audio.
-void ao_resume(struct ao *ao)
-{
-    if (ao->api->resume)
-        ao->api->resume(ao);
-}
-
-// Block until the current audio buffer has played completely.
-void ao_drain(struct ao *ao)
-{
-    if (ao->api->drain)
-        ao->api->drain(ao);
-}
-
-bool ao_eof_reached(struct ao *ao)
-{
-    return ao->api->get_eof ? ao->api->get_eof(ao) : true;
-}
-
 // Query the AO_EVENT_*s as requested by the events parameter, and return them.
 int ao_query_and_reset_events(struct ao *ao, int events)
 {
diff --git a/audio/out/buffer.c b/audio/out/buffer.c
new file mode 100644
index 0000000000..d890497113
--- /dev/null
+++ b/audio/out/buffer.c
@@ -0,0 +1,751 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stddef.h>
+#include <pthread.h>
+#include <inttypes.h>
+#include <unistd.h>
+#include <errno.h>
+#include <assert.h>
+
+#include "ao.h"
+#include "internal.h"
+#include "audio/aframe.h"
+#include "audio/format.h"
+
+#include "common/msg.h"
+#include "common/common.h"
+
+#include "input/input.h"
+
+#include "osdep/io.h"
+#include "osdep/timer.h"
+#include "osdep/threads.h"
+#include "osdep/atomic.h"
+#include "misc/ring.h"
+
+struct buffer_state {
+    pthread_mutex_t lock;
+    pthread_cond_t wakeup;
+
+    // Access from AO driver's thread only.
+    char *convert_buffer;
+
+    // --- protected by lock
+
+    struct mp_ring *buffers[MP_NUM_CHANNELS];
+
+
+    bool streaming;             // AO streaming active
+    bool playing;               // logically playing audio from buffer
+    bool paused;                // logically paused; implies playing=true
+    bool final_chunk;           // if buffer contains EOF
+
+    int64_t end_time_us;        // absolute output time of last played sample
+    int64_t underflow;          // number of samples missing since last check
+
+    bool need_wakeup;
+    bool initial_unblocked;
+
+    // "Push" AOs only (AOs with driver->play).
+    bool still_playing;
+    double expected_end_time;
+    bool wait_on_ao;
+    pthread_t thread;           // thread shoveling data to AO
+    bool thread_valid;          // thread is running
+    bool terminate;             // exit thread
+    struct mp_aframe *temp_buf;
+
+    int wakeup_pipe[2];
+};
+
+static void *playthread(void *arg);
+
+// lock must be held
+static void wakeup_playthread(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+    if (ao->driver->wakeup)
+        ao->driver->wakeup(ao);
+    p->need_wakeup = true;
+    pthread_cond_signal(&p->wakeup);
+}
+
+static int unlocked_get_space(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    int space = mp_ring_available(p->buffers[0]) / ao->sstride;
+
+    // The following code attempts to keep the total buffered audio at
+    // ao->buffer in order to improve latency.
+    if (ao->driver->play && ao->driver->get_space) {
+        int align = af_format_sample_alignment(ao->format);
+        int device_space = ao->driver->get_space(ao);
+        int device_buffered = ao->device_buffer - device_space;
+        int soft_buffered = mp_ring_size(p->buffers[0]) / ao->sstride - space;
+        // The extra margin helps avoiding too many wakeups if the AO is fully
+        // byte based and doesn't do proper chunked processing.
+        int min_buffer = ao->buffer + 64;
+        int missing = min_buffer - device_buffered - soft_buffered;
+        missing = (missing + align - 1) / align * align;
+        // But always keep the device's buffer filled as much as we can.
+        int device_missing = device_space - soft_buffered;
+        missing = MPMAX(missing, device_missing);
+        space = MPMIN(space, missing);
+        space = MPMAX(0, space);
+    }
+
+    return space;
+}
+
+// Return free size of the internal audio buffer. This controls how much audio
+// the core should decode and try to queue with ao_play().
+int ao_get_space(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+    pthread_mutex_lock(&p->lock);
+    int space = unlocked_get_space(ao);
+    pthread_mutex_unlock(&p->lock);
+    return space;
+}
+
+// Queue the given audio data. Start playback if it hasn't started yet. Return
+// the number of samples that was accepted (the core will try to queue the rest
+// again later). Should never block.
+//  data: start pointer for each plane. If the audio data is packed, only
+//        data[0] is valid, otherwise there is a plane for each channel.
+//  samples: size of the audio data (see ao->sstride)
+//  flags: currently AOPLAY_FINAL_CHUNK can be set
+int ao_play(struct ao *ao, void **data, int samples, int flags)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+
+    int write_samples = mp_ring_available(p->buffers[0]) / ao->sstride;
+    write_samples = MPMIN(write_samples, samples);
+
+    int write_bytes = write_samples * ao->sstride;
+    for (int n = 0; n < ao->num_planes; n++) {
+        int r = mp_ring_write(p->buffers[n], data[n], write_bytes);
+        assert(r == write_bytes);
+    }
+
+    p->paused = false;
+    p->final_chunk = write_samples == samples && (flags & AOPLAY_FINAL_CHUNK);
+
+    if (p->underflow)
+        MP_DBG(ao, "Audio underrun by %lld samples.\n", (long long)p->underflow);
+    p->underflow = 0;
+
+    if (write_samples) {
+        p->playing = true;
+        p->still_playing = true;
+        p->expected_end_time = 0;
+
+        if (!ao->driver->play && !p->streaming) {
+            p->streaming = true;
+            ao->driver->resume(ao);
+        }
+
+        wakeup_playthread(ao);
+    }
+    pthread_mutex_unlock(&p->lock);
+
+    return write_samples;
+}
+
+// Read the given amount of samples in the user-provided data buffer. Returns
+// the number of samples copied. If there is not enough data (buffer underrun
+// or EOF), return the number of samples that could be copied, and fill the
+// rest of the user-provided buffer with silence.
+// This basically assumes that the audio device doesn't care about underruns.
+// If this is called in paused mode, it will always return 0.
+// The caller should set out_time_us to the expected delay until the last sample
+// reaches the speakers, in microseconds, using mp_time_us() as reference.
+int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us)
+{
+    struct buffer_state *p = ao->buffer_state;
+    int full_bytes = samples * ao->sstride;
+    bool need_wakeup = false;
+    int bytes = 0;
+
+    pthread_mutex_lock(&p->lock);
+
+    if (!p->playing || p->paused)
+        goto end;
+
+    int buffered_bytes = mp_ring_buffered(p->buffers[0]);
+    bytes = MPMIN(buffered_bytes, full_bytes);
+
+    if (full_bytes > bytes && !p->final_chunk) {
+        p->underflow += (full_bytes - bytes) / ao->sstride;
+        ao_underrun_event(ao);
+    }
+
+    if (bytes > 0)
+        p->end_time_us = out_time_us;
+
+    for (int n = 0; n < ao->num_planes; n++)
+        mp_ring_read(p->buffers[n], data[n], bytes);
+
+    // Half of the buffer played -> request more.
+    need_wakeup = buffered_bytes - bytes <= mp_ring_size(p->buffers[0]) / 2;
+
+end:
+
+    pthread_mutex_unlock(&p->lock);
+
+    if (need_wakeup)
+        ao->wakeup_cb(ao->wakeup_ctx);
+
+    // pad with silence (underflow/paused/eof)
+    for (int n = 0; n < ao->num_planes; n++)
+        af_fill_silence((char *)data[n] + bytes, full_bytes - bytes, ao->format);
+
+    ao_post_process_data(ao, data, samples);
+
+    return bytes / ao->sstride;
+}
+
+// Same as ao_read_data(), but convert data according to *fmt.
+// fmt->src_fmt and fmt->channels must be the same as the AO parameters.
+int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
+                           void **data, int samples, int64_t out_time_us)
+{
+    struct buffer_state *p = ao->buffer_state;
+    void *ndata[MP_NUM_CHANNELS] = {0};
+
+    if (!ao_need_conversion(fmt))
+        return ao_read_data(ao, data, samples, out_time_us);
+
+    assert(ao->format == fmt->src_fmt);
+    assert(ao->channels.num == fmt->channels);
+
+    bool planar = af_fmt_is_planar(fmt->src_fmt);
+    int planes = planar ? fmt->channels : 1;
+    int plane_samples = samples * (planar ? 1: fmt->channels);
+    int src_plane_size = plane_samples * af_fmt_to_bytes(fmt->src_fmt);
+    int dst_plane_size = plane_samples * fmt->dst_bits / 8;
+
+    int needed = src_plane_size * planes;
+    if (needed > talloc_get_size(p->convert_buffer) || !p->convert_buffer) {
+        talloc_free(p->convert_buffer);
+        p->convert_buffer = talloc_size(NULL, needed);
+    }
+
+    for (int n = 0; n < planes; n++)
+        ndata[n] = p->convert_buffer + n * src_plane_size;
+
+    int res = ao_read_data(ao, ndata, samples, out_time_us);
+
+    ao_convert_inplace(fmt, ndata, samples);
+    for (int n = 0; n < planes; n++)
+        memcpy(data[n], ndata[n], dst_plane_size);
+
+    return res;
+}
+
+int ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct buffer_state *p = ao->buffer_state;
+    int r = CONTROL_UNKNOWN;
+    if (ao->driver->control) {
+        pthread_mutex_lock(&p->lock);
+        r = ao->driver->control(ao, cmd, arg);
+        pthread_mutex_unlock(&p->lock);
+    }
+    return r;
+}
+
+static double unlocked_get_delay(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+    double driver_delay = 0;
+
+    if (ao->driver->get_delay)
+        driver_delay = ao->driver->get_delay(ao);
+
+    if (!ao->driver->play) {
+        int64_t end = p->end_time_us;
+        int64_t now = mp_time_us();
+        driver_delay += MPMAX(0, (end - now) / (1000.0 * 1000.0));
+    }
+
+    return mp_ring_buffered(p->buffers[0]) / (double)ao->bps + driver_delay;
+}
+
+// Return size of the buffered data in seconds. Can include the device latency.
+// Basically, this returns how much data there is still to play, and how long
+// it takes until the last sample in the buffer reaches the speakers. This is
+// used for audio/video synchronization, so it's very important to implement
+// this correctly.
+double ao_get_delay(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+    double delay = unlocked_get_delay(ao);
+    pthread_mutex_unlock(&p->lock);
+    return delay;
+}
+
+// Stop playback and empty buffers. Essentially go back to the state after
+// ao->init().
+void ao_reset(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+
+    for (int n = 0; n < ao->num_planes; n++)
+        mp_ring_reset(p->buffers[n]);
+
+    if (!ao->stream_silence && ao->driver->reset) {
+        ao->driver->reset(ao); // assumes the audio callback thread is stopped
+        p->streaming = false;
+    }
+    p->paused = false;
+    p->playing = false;
+    if (p->still_playing)
+        wakeup_playthread(ao);
+    p->still_playing = false;
+    p->end_time_us = 0;
+
+    atomic_fetch_and(&ao->events_, ~(unsigned int)AO_EVENT_UNDERRUN);
+
+    pthread_mutex_unlock(&p->lock);
+}
+
+// Pause playback. Keep the current buffer. ao_get_delay() must return the
+// same value as before pausing.
+void ao_pause(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+
+    if (p->playing && !p->paused) {
+        if (p->streaming && !ao->stream_silence) {
+            if (ao->driver->pause) {
+                ao->driver->pause(ao);
+            } else if (ao->driver->reset) {
+                ao->driver->reset(ao);
+                p->streaming = false;
+            }
+        }
+        p->paused = true;
+        wakeup_playthread(ao);
+    }
+
+    pthread_mutex_unlock(&p->lock);
+}
+
+// Resume playback. Play the remaining buffer. If the driver doesn't support
+// pausing, it has to work around this and e.g. use ao_play_silence() to fill
+// the lost audio.
+void ao_resume(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+
+    if (p->playing && p->paused) {
+        if (p->streaming && ao->driver->resume)
+            ao->driver->resume(ao);
+        p->paused = false;
+        p->expected_end_time = 0;
+        wakeup_playthread(ao);
+    }
+
+    pthread_mutex_unlock(&p->lock);
+}
+
+bool ao_eof_reached(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+    bool eof = !p->playing;
+    if (ao->driver->play) {
+        eof |= !p->still_playing;
+    } else {
+        // For simplicity, ignore the latency. Otherwise, we would have to run
+        // an extra thread to time it.
+        eof |= mp_ring_buffered(p->buffers[0]) == 0;
+    }
+    pthread_mutex_unlock(&p->lock);
+
+    return eof;
+}
+
+// Block until the current audio buffer has played completely.
+void ao_drain(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    pthread_mutex_lock(&p->lock);
+    p->final_chunk = true;
+    wakeup_playthread(ao);
+    double left = 0;
+    if (p->playing && !p->paused)
+        left = mp_ring_buffered(p->buffers[0]) / (double)ao->bps * 1e6;
+    pthread_mutex_unlock(&p->lock);
+
+    // Wait for lower bound.
+    mp_sleep_us(left);
+    // And then poll for actual end. (Unfortunately, this code considers
+    // audio APIs which do not want you to use mutexes in the audio
+    // callback, and an extra semaphore would require slightly more effort.)
+    // Limit to arbitrary ~250ms max. waiting for robustness.
+    int64_t max = mp_time_us() + 250000;
+    while (mp_time_us() < max && !ao_eof_reached(ao))
+        mp_sleep_us(1);
+
+    ao_reset(ao);
+}
+
+// Uninitialize and destroy the AO. Remaining audio must be dropped.
+void ao_uninit(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    if (p->thread_valid) {
+        pthread_mutex_lock(&p->lock);
+        p->terminate = true;
+        wakeup_playthread(ao);
+        pthread_mutex_unlock(&p->lock);
+
+        pthread_join(p->thread, NULL);
+        p->thread_valid = false;
+    }
+
+    if (ao->driver_initialized)
+        ao->driver->uninit(ao);
+
+    talloc_free(p->convert_buffer);
+    talloc_free(p->temp_buf);
+
+    for (int n = 0; n < 2; n++) {
+        int h = p->wakeup_pipe[n];
+        if (h >= 0)
+            close(h);
+    }
+
+    pthread_cond_destroy(&p->wakeup);
+    pthread_mutex_destroy(&p->lock);
+
+    talloc_free(ao);
+}
+
+void init_buffer_pre(struct ao *ao)
+{
+    ao->buffer_state = talloc_zero(ao, struct buffer_state);
+}
+
+bool init_buffer_post(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    if (!ao->driver->play)
+        assert(ao->driver->resume);
+
+    for (int n = 0; n < ao->num_planes; n++)
+        p->buffers[n] = mp_ring_new(ao, ao->buffer * ao->sstride);
+
+    mpthread_mutex_init_recursive(&p->lock);
+    pthread_cond_init(&p->wakeup, NULL);
+    mp_make_wakeup_pipe(p->wakeup_pipe);
+
+    if (ao->driver->play) {
+        if (ao->device_buffer <= 0) {
+            MP_FATAL(ao, "Couldn't probe device buffer size.\n");
+            return false;
+        }
+
+        p->thread_valid = true;
+        if (pthread_create(&p->thread, NULL, playthread, ao)) {
+            p->thread_valid = false;
+            return false;
+        }
+    } else {
+        if (ao->stream_silence) {
+            ao->driver->resume(ao);
+            p->streaming = true;
+        }
+    }
+
+    return true;
+}
+
+static bool realloc_buf(struct ao *ao, int samples)
+{
+    struct buffer_state *p = ao->buffer_state;
+
+    samples = MPMAX(1, samples);
+
+    if (!p->temp_buf || samples > mp_aframe_get_size(p->temp_buf)) {
+        TA_FREEP(&p->temp_buf);
+        p->temp_buf = mp_aframe_create();
+        if (!mp_aframe_set_format(p->temp_buf, ao->format) ||
+            !mp_aframe_set_chmap(p->temp_buf, &ao->channels) ||
+            !mp_aframe_set_rate(p->temp_buf, ao->samplerate) ||
+            !mp_aframe_alloc_data(p->temp_buf, samples))
+        {
+            TA_FREEP(&p->temp_buf);
+            return false;
+        }
+    }
+
+    return true;
+}
+
+// called locked
+static void ao_play_data(struct ao *ao)
+{
+    struct buffer_state *p = ao->buffer_state;
+    int space = ao->driver->get_space(ao);
+    bool play_silence = p->paused || (ao->stream_silence && !p->still_playing);
+    space = MPMAX(space, 0);
+    // Most AOs want period-size aligned audio, and preferably as much as
+    // possible in one go, so the audio data is "linearized" into this buffer.
+    if (space % ao->period_size)
+        MP_ERR(ao, "Audio device reports unaligned available buffer size.\n");
+    if (!realloc_buf(ao, space)) {
+        MP_ERR(ao, "Failed to allocate buffer.\n");
+        return;
+    }
+    void **planes = (void **)mp_aframe_get_data_rw(p->temp_buf);
+    assert(planes);
+    int samples = mp_ring_buffered(p->buffers[0]) / ao->sstride;
+    if (samples > space)
+        samples = space;
+    if (play_silence)
+        samples = space;
+    samples = ao_read_data(ao, planes, samples, 0);
+    if (play_silence)
+        samples = space; // ao_read_data() sets remainder to silent
+    int max = samples;
+    int flags = 0;
+    if (p->final_chunk && samples < space) {
+        flags |= AOPLAY_FINAL_CHUNK;
+    } else {
+        samples = samples / ao->period_size * ao->period_size;
+    }
+    MP_STATS(ao, "start ao fill");
+    ao_post_process_data(ao, planes, samples);
+    int r = 0;
+    if (samples)
+        r = ao->driver->play(ao, planes, samples, flags);
+    MP_STATS(ao, "end ao fill");
+    if (r > samples) {
+        MP_ERR(ao, "Audio device returned nonsense value.\n");
+        r = samples;
+    } else if (r < 0) {
+        MP_ERR(ao, "Error writing audio to device.\n");
+    } else if (r != samples) {
+        MP_ERR(ao, "Audio device returned broken buffer state (sent %d samples, "
+               "got %d samples, %d period%s)! Discarding audio.\n", samples, r,
+               ao->period_size, flags & AOPLAY_FINAL_CHUNK ? " final" : "");
+    }
+    r = MPMAX(r, 0);
+    // Probably can't copy the rest of the buffer due to period alignment.
+    bool stuck_eof = r <= 0 && space >= max && samples > 0;
+    if ((flags & AOPLAY_FINAL_CHUNK) && stuck_eof) {
+        MP_ERR(ao, "Audio output driver seems to ignore AOPLAY_FINAL_CHUNK.\n");
+        r = max;
+    }
+    if (r > 0) {
+        p->expected_end_time = 0;
+        p->streaming = true;
+    }
+    // Nothing written, but more input data than space - this must mean the
+    // AO's get_space() doesn't do period alignment correctly.
+    bool stuck = r == 0 && max >= space && space > 0;
+    if (stuck)
+        MP_ERR(ao, "Audio output is reporting incorrect buffer status.\n");
+    // Wait until space becomes available. Also wait if we actually wrote data,
+    // so the AO wakes us up properly if it needs more data.
+    p->wait_on_ao = space == 0 || r > 0 || stuck;
+    p->still_playing |= r > 0 && !play_silence;
+    // If we just filled the AO completely (r == space), don't refill for a
+    // while. Prevents wakeup feedback with byte-granular AOs.
+    int needed = unlocked_get_space(ao);
+    bool more = needed >= (r == space ? ao->device_buffer / 4 : 1) && !stuck &&
+                !(flags & AOPLAY_FINAL_CHUNK);
+    if (more)
+        ao->wakeup_cb(ao->wakeup_ctx); // request more data
+    if (!samples && space && !ao->driver->reports_underruns && p->still_playing)
+        ao_underrun_event(ao);
+    MP_TRACE(ao, "in=%d flags=%d space=%d r=%d wa/pl=%d/%d needed=%d more=%d\n",
+             max, flags, space, r, p->wait_on_ao, p->still_playing, needed, more);
+}
+
+static void *playthread(void *arg)
+{
+    struct ao *ao = arg;
+    struct buffer_state *p = ao->buffer_state;
+    mpthread_set_name("ao");
+    pthread_mutex_lock(&p->lock);
+    while (!p->terminate) {
+        bool blocked = ao->driver->initially_blocked && !p->initial_unblocked;
+        bool playing = (!p->paused || ao->stream_silence) && !blocked;
+        if (playing)
+            ao_play_data(ao);
+
+        if (!p->need_wakeup) {
+            MP_STATS(ao, "start audio wait");
+            if (!p->wait_on_ao || !playing) {
+                // Avoid busy waiting, because the audio API will still report
+                // that it needs new data, even if we're not ready yet, or if
+                // get_space() decides that the amount of audio buffered in the
+                // device is enough, and p->buffer can be empty.
+                // The most important part is that the decoder is woken up, so
+                // that the decoder will wake up us in turn.
+                MP_TRACE(ao, "buffer inactive.\n");
+
+                bool was_playing = p->still_playing;
+                double timeout = -1;
+                if (p->still_playing && !p->paused && p->final_chunk &&
+                    !mp_ring_buffered(p->buffers[0]))
+                {
+                    double now = mp_time_sec();
+                    if (!p->expected_end_time)
+                        p->expected_end_time = now + unlocked_get_delay(ao);
+                    if (p->expected_end_time < now) {
+                        p->still_playing = false;
+                    } else {
+                        timeout = p->expected_end_time - now;
+                    }
+                }
+
+                if (was_playing && !p->still_playing)
+                    ao->wakeup_cb(ao->wakeup_ctx);
+                pthread_cond_signal(&p->wakeup); // for draining
+
+                if (p->still_playing && timeout > 0) {
+                    struct timespec ts = mp_rel_time_to_timespec(timeout);
+                    pthread_cond_timedwait(&p->wakeup, &p->lock, &ts);
+                } else {
+                    pthread_cond_wait(&p->wakeup, &p->lock);
+                }
+            } else {
+                // Wait until the device wants us to write more data to it.
+                if (!ao->driver->wait || ao->driver->wait(ao, &p->lock) < 0) {
+                    // Fallback to guessing.
+                    double timeout = 0;
+                    if (ao->driver->get_delay)
+                        timeout = ao->driver->get_delay(ao);
+                    timeout *= 0.25; // wake up if 25% played
+                    if (!p->need_wakeup) {
+                        struct timespec ts = mp_rel_time_to_timespec(timeout);
+                        pthread_cond_timedwait(&p->wakeup, &p->lock, &ts);
+                    }
+                }
+            }
+            MP_STATS(ao, "end audio wait");
+        }
+        p->need_wakeup = false;
+    }
+    pthread_mutex_unlock(&p->lock);
+    return NULL;
+}
+
+void ao_unblock(struct ao *ao)
+{
+    if (ao->driver->play) {
+        struct buffer_state *p = ao->buffer_state;
+        pthread_mutex_lock(&p->lock);
+        p->need_wakeup = true;
+        p->initial_unblocked = true;
+        wakeup_playthread(ao);
+        pthread_cond_signal(&p->wakeup);
+        pthread_mutex_unlock(&p->lock);
+    }
+}
+
+// Must be called locked.
+int ao_play_silence(struct ao *ao, int samples)
+{
+    assert(ao->driver->play);
+
+    struct buffer_state *p = ao->buffer_state;
+
+    if (!realloc_buf(ao, samples) || !ao->driver->play)
+        return 0;
+
+    void **planes = (void **)mp_aframe_get_data_rw(p->temp_buf);
+    assert(planes);
+
+    for (int n = 0; n < ao->num_planes; n++)
+        af_fill_silence(planes[n], ao->sstride * samples, ao->format);
+
+    return ao->driver->play(ao, planes, samples, 0);
+}
+
+#ifndef __MINGW32__
+
+#include <poll.h>
+
+#define MAX_POLL_FDS 20
+
+// Call poll() for the given fds. This will extend the given fds with the
+// wakeup pipe, so ao_wakeup_poll() will basically interrupt this function.
+// Unlocks the lock temporarily.
+// Returns <0 on error, 0 on success, 1 if the caller should return immediately.
+int ao_wait_poll(struct ao *ao, struct pollfd *fds, int num_fds,
+                 pthread_mutex_t *lock)
+{
+    struct buffer_state *p = ao->buffer_state;
+    assert(ao->driver->play);
+    assert(&p->lock == lock);
+
+    if (num_fds >= MAX_POLL_FDS || p->wakeup_pipe[0] < 0)
+        return -1;
+
+    struct pollfd p_fds[MAX_POLL_FDS];
+    memcpy(p_fds, fds, num_fds * sizeof(p_fds[0]));
+    p_fds[num_fds] = (struct pollfd){
+        .fd = p->wakeup_pipe[0],
+        .events = POLLIN,
+    };
+
+    pthread_mutex_unlock(&p->lock);
+    int r = poll(p_fds, num_fds + 1, -1);
+    r = r < 0 ? -errno : 0;
+    pthread_mutex_lock(&p->lock);
+
+    memcpy(fds, p_fds, num_fds * sizeof(fds[0]));
+    bool wakeup = false;
+    if (p_fds[num_fds].revents & POLLIN) {
+        wakeup = true;
+        // might "drown" some wakeups, but that's ok for our use-case
+        mp_flush_wakeup_pipe(p->wakeup_pipe[0]);
+    }
+    return (r >= 0 || r == -EINTR) ? wakeup : -1;
+}
+
+void ao_wakeup_poll(struct ao *ao)
+{
+    assert(ao->driver->play);
+    struct buffer_state *p = ao->buffer_state;
+
+    (void)write(p->wakeup_pipe[1], &(char){0}, 1);
+}
+
+#endif
diff --git a/audio/out/internal.h b/audio/out/internal.h
index 7bdd17ad8a..c036ebad16 100644
--- a/audio/out/internal.h
+++ b/audio/out/internal.h
@@ -37,8 +37,8 @@ struct ao {
     bool untimed;               // don't assume realtime playback
     int device_buffer;          // device buffer in samples (guessed by
                                 // common init code if not set by driver)
-    const struct ao_driver *api; // entrypoints to the wrapper (push.c/pull.c)
     const struct ao_driver *driver;
+    bool driver_initialized;
     void *priv;
     struct mpv_global *global;
     struct encode_lavc_context *encode_lavc_ctx;
@@ -75,12 +75,12 @@ struct ao {
 
     int buffer;
     double def_buffer;
+    struct buffer_state *buffer_state;
     void *api_priv;
 };
 
-extern const struct ao_driver ao_api_push;
-extern const struct ao_driver ao_api_pull;
-
+void init_buffer_pre(struct ao *ao);
+bool init_buffer_post(struct ao *ao);
 
 /* Note:
  *
diff --git a/audio/out/pull.c b/audio/out/pull.c
deleted file mode 100644
index fe8204e01b..0000000000
--- a/audio/out/pull.c
+++ /dev/null
@@ -1,340 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stddef.h>
-#include <inttypes.h>
-#include <assert.h>
-
-#include "ao.h"
-#include "internal.h"
-#include "audio/format.h"
-
-#include "common/msg.h"
-#include "common/common.h"
-
-#include "input/input.h"
-
-#include "osdep/timer.h"
-#include "osdep/threads.h"
-#include "osdep/atomic.h"
-#include "misc/ring.h"
-
-/*
- * Note: there is some stupid stuff in this file in order to avoid mutexes.
- * This requirement is dictated by several audio APIs, at least jackaudio.
- */
-
-enum {
-    AO_STATE_NONE,  // idle (e.g. before playback started, or after playback
-                    // finished, but device is open)
-    AO_STATE_WAIT,  // wait for callback to go into AO_STATE_NONE state
-    AO_STATE_PLAY,  // play the buffer
-    AO_STATE_BUSY,  // like AO_STATE_PLAY, but ao_read_data() is being called
-};
-
-#define IS_PLAYING(st) ((st) == AO_STATE_PLAY || (st) == AO_STATE_BUSY)
-
-struct ao_pull_state {
-    // Be very careful with the order when accessing planes.
-    struct mp_ring *buffers[MP_NUM_CHANNELS];
-
-    // AO_STATE_*
-    atomic_int state;
-
-    // Set when the buffer is intentionally not fed anymore in PLAY state.
-    atomic_bool draining;
-
-    // Set by the audio thread when an underflow was detected.
-    // It adds the number of samples.
-    atomic_int underflow;
-
-    // Device delay of the last written sample, in realtime.
-    atomic_llong end_time_us;
-
-    char *convert_buffer;
-};
-
-static void set_state(struct ao *ao, int new_state)
-{
-    struct ao_pull_state *p = ao->api_priv;
-    while (1) {
-        int old = atomic_load(&p->state);
-        if (old == AO_STATE_BUSY) {
-            // A spinlock, because some audio APIs don't want us to use mutexes.
-            mp_sleep_us(1);
-            continue;
-        }
-        if (atomic_compare_exchange_strong(&p->state, &old, new_state))
-            break;
-    }
-}
-
-static int get_space(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-
-    // Since the reader will read the last plane last, its free space is the
-    // minimum free space across all planes.
-    return mp_ring_available(p->buffers[ao->num_planes - 1]) / ao->sstride;
-}
-
-static int play(struct ao *ao, void **data, int samples, int flags)
-{
-    struct ao_pull_state *p = ao->api_priv;
-
-    int write_samples = get_space(ao);
-    write_samples = MPMIN(write_samples, samples);
-
-    // Write starting from the last plane - this way, the first plane will
-    // always contain the minimum amount of data readable across all planes
-    // (assumes the reader starts with the first plane).
-    int write_bytes = write_samples * ao->sstride;
-    for (int n = ao->num_planes - 1; n >= 0; n--) {
-        int r = mp_ring_write(p->buffers[n], data[n], write_bytes);
-        assert(r == write_bytes);
-    }
-
-    int state = atomic_load(&p->state);
-    if (!IS_PLAYING(state)) {
-        atomic_store(&p->draining, false);
-        atomic_store(&p->underflow, 0);
-        set_state(ao, AO_STATE_PLAY);
-        if (!ao->stream_silence)
-            ao->driver->resume(ao);
-    }
-
-    bool draining = write_samples == samples && (flags & AOPLAY_FINAL_CHUNK);
-    atomic_store(&p->draining, draining);
-
-    int underflow = atomic_fetch_and(&p->underflow, 0);
-    if (underflow)
-        MP_DBG(ao, "Audio underrun by %d samples.\n", underflow);
-
-    return write_samples;
-}
-
-// Read the given amount of samples in the user-provided data buffer. Returns
-// the number of samples copied. If there is not enough data (buffer underrun
-// or EOF), return the number of samples that could be copied, and fill the
-// rest of the user-provided buffer with silence.
-// This basically assumes that the audio device doesn't care about underruns.
-// If this is called in paused mode, it will always return 0.
-// The caller should set out_time_us to the expected delay until the last sample
-// reaches the speakers, in microseconds, using mp_time_us() as reference.
-int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us)
-{
-    assert(ao->api == &ao_api_pull);
-
-    struct ao_pull_state *p = ao->api_priv;
-    int full_bytes = samples * ao->sstride;
-    bool need_wakeup = false;
-    int bytes = 0;
-
-    // Play silence in states other than AO_STATE_PLAY.
-    if (!atomic_compare_exchange_strong(&p->state, &(int){AO_STATE_PLAY},
-                                        AO_STATE_BUSY))
-        goto end;
-
-    // Since the writer will write the first plane last, its buffered amount
-    // of data is the minimum amount across all planes.
-    int buffered_bytes = mp_ring_buffered(p->buffers[0]);
-    bytes = MPMIN(buffered_bytes, full_bytes);
-
-    if (full_bytes > bytes && !atomic_load(&p->draining)) {
-        atomic_fetch_add(&p->underflow, (full_bytes - bytes) / ao->sstride);
-        ao_underrun_event(ao);
-    }
-
-    if (bytes > 0)
-        atomic_store(&p->end_time_us, out_time_us);
-
-    for (int n = 0; n < ao->num_planes; n++) {
-        int r = mp_ring_read(p->buffers[n], data[n], bytes);
-        bytes = MPMIN(bytes, r);
-    }
-
-    // Half of the buffer played -> request more.
-    need_wakeup = buffered_bytes - bytes <= mp_ring_size(p->buffers[0]) / 2;
-
-    // Should never fail.
-    atomic_compare_exchange_strong(&p->state, &(int){AO_STATE_BUSY}, AO_STATE_PLAY);
-
-end:
-
-    if (need_wakeup)
-        ao->wakeup_cb(ao->wakeup_ctx);
-
-    // pad with silence (underflow/paused/eof)
-    for (int n = 0; n < ao->num_planes; n++)
-        af_fill_silence((char *)data[n] + bytes, full_bytes - bytes, ao->format);
-
-    ao_post_process_data(ao, data, samples);
-
-    return bytes / ao->sstride;
-}
-
-// Same as ao_read_data(), but convert data according to *fmt.
-// fmt->src_fmt and fmt->channels must be the same as the AO parameters.
-int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
-                           void **data, int samples, int64_t out_time_us)
-{
-    assert(ao->api == &ao_api_pull);
-
-    struct ao_pull_state *p = ao->api_priv;
-    void *ndata[MP_NUM_CHANNELS] = {0};
-
-    if (!ao_need_conversion(fmt))
-        return ao_read_data(ao, data, samples, out_time_us);
-
-    assert(ao->format == fmt->src_fmt);
-    assert(ao->channels.num == fmt->channels);
-
-    bool planar = af_fmt_is_planar(fmt->src_fmt);
-    int planes = planar ? fmt->channels : 1;
-    int plane_samples = samples * (planar ? 1: fmt->channels);
-    int src_plane_size = plane_samples * af_fmt_to_bytes(fmt->src_fmt);
-    int dst_plane_size = plane_samples * fmt->dst_bits / 8;
-
-    int needed = src_plane_size * planes;
-    if (needed > talloc_get_size(p->convert_buffer) || !p->convert_buffer) {
-        talloc_free(p->convert_buffer);
-        p->convert_buffer = talloc_size(NULL, needed);
-    }
-
-    for (int n = 0; n < planes; n++)
-        ndata[n] = p->convert_buffer + n * src_plane_size;
-
-    int res = ao_read_data(ao, ndata, samples, out_time_us);
-
-    ao_convert_inplace(fmt, ndata, samples);
-    for (int n = 0; n < planes; n++)
-        memcpy(data[n], ndata[n], dst_plane_size);
-
-    return res;
-}
-
-static int control(struct ao *ao, enum aocontrol cmd, void *arg)
-{
-    if (ao->driver->control)
-        return ao->driver->control(ao, cmd, arg);
-    return CONTROL_UNKNOWN;
-}
-
-// Return size of the buffered data in seconds. Can include the device latency.
-// Basically, this returns how much data there is still to play, and how long
-// it takes until the last sample in the buffer reaches the speakers. This is
-// used for audio/video synchronization, so it's very important to implement
-// this correctly.
-static double get_delay(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-
-    int64_t end = atomic_load(&p->end_time_us);
-    int64_t now = mp_time_us();
-    double driver_delay = MPMAX(0, (end - now) / (1000.0 * 1000.0));
-    return mp_ring_buffered(p->buffers[0]) / (double)ao->bps + driver_delay;
-}
-
-static void reset(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-    if (!ao->stream_silence && ao->driver->reset)
-        ao->driver->reset(ao); // assumes the audio callback thread is stopped
-    set_state(ao, AO_STATE_NONE);
-    for (int n = 0; n < ao->num_planes; n++)
-        mp_ring_reset(p->buffers[n]);
-    atomic_store(&p->end_time_us, 0);
-}
-
-static void pause(struct ao *ao)
-{
-    if (!ao->stream_silence && ao->driver->reset)
-        ao->driver->reset(ao);
-    set_state(ao, AO_STATE_NONE);
-}
-
-static void resume(struct ao *ao)
-{
-    set_state(ao, AO_STATE_PLAY);
-    if (!ao->stream_silence)
-        ao->driver->resume(ao);
-}
-
-static bool get_eof(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-    // For simplicity, ignore the latency. Otherwise, we would have to run an
-    // extra thread to time it.
-    return mp_ring_buffered(p->buffers[0]) == 0;
-}
-
-static void drain(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-    int state = atomic_load(&p->state);
-    if (IS_PLAYING(state)) {
-        atomic_store(&p->draining, true);
-        // Wait for lower bound.
-        mp_sleep_us(mp_ring_buffered(p->buffers[0]) / (double)ao->bps * 1e6);
-        // And then poll for actual end. (Unfortunately, this code considers
-        // audio APIs which do not want you to use mutexes in the audio
-        // callback, and an extra semaphore would require slightly more effort.)
-        // Limit to arbitrary ~250ms max. waiting for robustness.
-        int64_t max = mp_time_us() + 250000;
-        while (mp_time_us() < max && !get_eof(ao))
-            mp_sleep_us(1);
-    }
-    reset(ao);
-}
-
-static void uninit(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-
-    ao->driver->uninit(ao);
-
-    talloc_free(p->convert_buffer);
-}
-
-static int init(struct ao *ao)
-{
-    struct ao_pull_state *p = ao->api_priv;
-    for (int n = 0; n < ao->num_planes; n++)
-        p->buffers[n] = mp_ring_new(ao, ao->buffer * ao->sstride);
-    atomic_store(&p->state, AO_STATE_NONE);
-    assert(ao->driver->resume);
-
-    if (ao->stream_silence)
-        ao->driver->resume(ao);
-
-    return 0;
-}
-
-const struct ao_driver ao_api_pull = {
-    .init = init,
-    .control = control,
-    .uninit = uninit,
-    .drain = drain,
-    .reset = reset,
-    .get_space = get_space,
-    .play = play,
-    .get_delay = get_delay,
-    .get_eof = get_eof,
-    .pause = pause,
-    .resume = resume,
-    .priv_size = sizeof(struct ao_pull_state),
-};
diff --git a/audio/out/push.c b/audio/out/push.c
deleted file mode 100644
index 92fd53631b..0000000000
--- a/audio/out/push.c
+++ /dev/null
@@ -1,572 +0,0 @@
-/*
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stddef.h>
-#include <pthread.h>
-#include <inttypes.h>
-#include <unistd.h>
-#include <errno.h>
-#include <assert.h>
-
-#include "osdep/io.h"
-
-#include "ao.h"
-#include "internal.h"
-#include "audio/format.h"
-
-#include "common/msg.h"
-#include "common/common.h"
-
-#include "input/input.h"
-
-#include "osdep/threads.h"
-#include "osdep/timer.h"
-#include "osdep/atomic.h"
-
-#include "audio/audio_buffer.h"
-
-struct ao_push_state {
-    pthread_t thread;
-    pthread_mutex_t lock;
-    pthread_cond_t wakeup;
-
-    // --- protected by lock
-
-    struct mp_audio_buffer *buffer;
-
-    uint8_t *silence[MP_NUM_CHANNELS];
-    int silence_samples;
-
-    bool terminate;
-    bool wait_on_ao;
-    bool still_playing;
-    bool need_wakeup;
-    bool paused;
-    bool initial_unblocked;
-
-    // Whether the current buffer contains the complete audio.
-    bool final_chunk;
-    double expected_end_time;
-
-    int wakeup_pipe[2];
-};
-
-// lock must be held
-static void wakeup_playthread(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    if (ao->driver->wakeup)
-        ao->driver->wakeup(ao);
-    p->need_wakeup = true;
-    pthread_cond_signal(&p->wakeup);
-}
-
-static int control(struct ao *ao, enum aocontrol cmd, void *arg)
-{
-    int r = CONTROL_UNKNOWN;
-    if (ao->driver->control) {
-        struct ao_push_state *p = ao->api_priv;
-        pthread_mutex_lock(&p->lock);
-        r = ao->driver->control(ao, cmd, arg);
-        pthread_mutex_unlock(&p->lock);
-    }
-    return r;
-}
-
-static double unlocked_get_delay(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    double driver_delay = 0;
-    if (ao->driver->get_delay)
-        driver_delay = ao->driver->get_delay(ao);
-    return driver_delay + mp_audio_buffer_seconds(p->buffer);
-}
-
-static double get_delay(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    pthread_mutex_lock(&p->lock);
-    double delay = unlocked_get_delay(ao);
-    pthread_mutex_unlock(&p->lock);
-    return delay;
-}
-
-static void reset(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    pthread_mutex_lock(&p->lock);
-    if (ao->driver->reset)
-        ao->driver->reset(ao);
-    mp_audio_buffer_clear(p->buffer);
-    p->paused = false;
-    if (p->still_playing)
-        wakeup_playthread(ao);
-    p->still_playing = false;
-    pthread_mutex_unlock(&p->lock);
-}
-
-static void audio_pause(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    pthread_mutex_lock(&p->lock);
-    if (ao->driver->pause)
-        ao->driver->pause(ao);
-    p->paused = true;
-    wakeup_playthread(ao);
-    pthread_mutex_unlock(&p->lock);
-}
-
-static void resume(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    pthread_mutex_lock(&p->lock);
-    if (ao->driver->resume)
-        ao->driver->resume(ao);
-    p->paused = false;
-    p->expected_end_time = 0;
-    wakeup_playthread(ao);
-    pthread_mutex_unlock(&p->lock);
-}
-
-static void drain(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    double maxbuffer = ao->buffer / (double)ao->samplerate + 1;
-
-    MP_VERBOSE(ao, "draining...\n");
-
-    pthread_mutex_lock(&p->lock);
-    if (p->paused)
-        goto done;
-
-    p->final_chunk = true;
-    wakeup_playthread(ao);
-
-    // Wait until everything is done. Since the audio API (especially ALSA)
-    // can't be trusted to do this right, and we're hard-blocking here, apply
-    // an upper bound timeout.
-    struct timespec until = mp_rel_time_to_timespec(maxbuffer);
-    while (p->still_playing && mp_audio_buffer_samples(p->buffer) > 0) {
-        if (pthread_cond_timedwait(&p->wakeup, &p->lock, &until)) {
-            MP_WARN(ao, "Draining is taking too long, aborting.\n");
-            goto done;
-        }
-    }
-
-    if (ao->driver->drain) {
-        ao->driver->drain(ao);
-    } else {
-        double time = unlocked_get_delay(ao);
-        mp_sleep_us(MPMIN(time, maxbuffer) * 1e6);
-    }
-
-done:
-    pthread_mutex_unlock(&p->lock);
-
-    reset(ao);
-}
-
-static int unlocked_get_space(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    int space = mp_audio_buffer_get_write_available(p->buffer);
-    if (ao->driver->get_space) {
-        int align = af_format_sample_alignment(ao->format);
-        // The following code attempts to keep the total buffered audio to
-        // ao->buffer in order to improve latency.
-        int device_space = ao->driver->get_space(ao);
-        int device_buffered = ao->device_buffer - device_space;
-        int soft_buffered = mp_audio_buffer_samples(p->buffer);
-        // The extra margin helps avoiding too many wakeups if the AO is fully
-        // byte based and doesn't do proper chunked processing.
-        int min_buffer = ao->buffer + 64;
-        int missing = min_buffer - device_buffered - soft_buffered;
-        missing = (missing + align - 1) / align * align;
-        // But always keep the device's buffer filled as much as we can.
-        int device_missing = device_space - soft_buffered;
-        missing = MPMAX(missing, device_missing);
-        space = MPMIN(space, missing);
-        space = MPMAX(0, space);
-    }
-    return space;
-}
-
-static int get_space(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    pthread_mutex_lock(&p->lock);
-    int space = unlocked_get_space(ao);
-    pthread_mutex_unlock(&p->lock);
-    return space;
-}
-
-static bool get_eof(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    pthread_mutex_lock(&p->lock);
-    bool eof = !p->still_playing;
-    pthread_mutex_unlock(&p->lock);
-    return eof;
-}
-
-static int play(struct ao *ao, void **data, int samples, int flags)
-{
-    struct ao_push_state *p = ao->api_priv;
-
-    pthread_mutex_lock(&p->lock);
-
-    int write_samples = mp_audio_buffer_get_write_available(p->buffer);
-    write_samples = MPMIN(write_samples, samples);
-
-    MP_TRACE(ao, "samples=%d flags=%d r=%d\n", samples, flags, write_samples);
-
-    if (write_samples < samples)
-        flags = flags & ~AOPLAY_FINAL_CHUNK;
-    bool is_final = flags & AOPLAY_FINAL_CHUNK;
-
-    mp_audio_buffer_append(p->buffer, data, samples);
-
-    bool got_data = write_samples > 0 || p->paused || p->final_chunk != is_final;
-
-    p->final_chunk = is_final;
-    p->paused = false;
-    if (got_data) {
-        p->still_playing = true;
-        p->expected_end_time = 0;
-
-        // If we don't have new data, the decoder thread basically promises it
-        // will send new data as soon as it's available.
-        wakeup_playthread(ao);
-    }
-    pthread_mutex_unlock(&p->lock);
-    return write_samples;
-}
-
-static bool realloc_silence(struct ao *ao, int samples)
-{
-    struct ao_push_state *p = ao->api_priv;
-
-    if (samples <= 0 || !af_fmt_is_pcm(ao->format))
-        return false;
-
-    if (samples > p->silence_samples) {
-        talloc_free(p->silence[0]);
-
-        int bytes = af_fmt_to_bytes(ao->format) * samples * ao->channels.num;
-        p->silence[0] = talloc_size(p, bytes);
-        for (int n = 1; n < MP_NUM_CHANNELS; n++)
-            p->silence[n] = p->silence[0];
-        p->silence_samples = samples;
-
-        af_fill_silence(p->silence[0], bytes, ao->format);
-    }
-
-    return true;
-}
-
-// called locked
-static void ao_play_data(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-    int space = ao->driver->get_space(ao);
-    bool play_silence = p->paused || (ao->stream_silence && !p->still_playing);
-    space = MPMAX(space, 0);
-    if (space % ao->period_size)
-        MP_ERR(ao, "Audio device reports unaligned available buffer size.\n");
-    uint8_t **planes;
-    int samples;
-    if (play_silence) {
-        planes = p->silence;
-        samples = realloc_silence(ao, space) ? space : 0;
-    } else {
-        mp_audio_buffer_peek(p->buffer, &planes, &samples);
-    }
-    int max = samples;
-    if (samples > space)
-        samples = space;
-    int flags = 0;
-    if (p->final_chunk && samples == max) {
-        flags |= AOPLAY_FINAL_CHUNK;
-    } else {
-        samples = samples / ao->period_size * ao->period_size;
-    }
-    MP_STATS(ao, "start ao fill");
-    ao_post_process_data(ao, (void **)planes, samples);
-    int r = 0;
-    if (samples)
-        r = ao->driver->play(ao, (void **)planes, samples, flags);
-    MP_STATS(ao, "end ao fill");
-    if (r > samples) {
-        MP_ERR(ao, "Audio device returned nonsense value.\n");
-        r = samples;
-    } else if (r < 0) {
-        MP_ERR(ao, "Error writing audio to device.\n");
-    } else if (r != samples) {
-        MP_ERR(ao, "Audio device returned broken buffer state (sent %d samples, "
-               "got %d samples, %d period%s)!\n", samples, r,
-               ao->period_size, flags & AOPLAY_FINAL_CHUNK ? " final" : "");
-    }
-    r = MPMAX(r, 0);
-    // Probably can't copy the rest of the buffer due to period alignment.
-    bool stuck_eof = r <= 0 && space >= max && samples > 0;
-    if ((flags & AOPLAY_FINAL_CHUNK) && stuck_eof) {
-        MP_ERR(ao, "Audio output driver seems to ignore AOPLAY_FINAL_CHUNK.\n");
-        r = max;
-    }
-    if (!play_silence)
-        mp_audio_buffer_skip(p->buffer, r);
-    if (r > 0)
-        p->expected_end_time = 0;
-    // Nothing written, but more input data than space - this must mean the
-    // AO's get_space() doesn't do period alignment correctly.
-    bool stuck = r == 0 && max >= space && space > 0;
-    if (stuck)
-        MP_ERR(ao, "Audio output is reporting incorrect buffer status.\n");
-    // Wait until space becomes available. Also wait if we actually wrote data,
-    // so the AO wakes us up properly if it needs more data.
-    p->wait_on_ao = space == 0 || r > 0 || stuck;
-    p->still_playing |= r > 0 && !play_silence;
-    // If we just filled the AO completely (r == space), don't refill for a
-    // while. Prevents wakeup feedback with byte-granular AOs.
-    int needed = unlocked_get_space(ao);
-    bool more = needed >= (r == space ? ao->device_buffer / 4 : 1) && !stuck &&
-                !(flags & AOPLAY_FINAL_CHUNK);
-    if (more)
-        ao->wakeup_cb(ao->wakeup_ctx); // request more data
-    if (!samples && space && !ao->driver->reports_underruns && p->still_playing)
-        ao_underrun_event(ao);
-    MP_TRACE(ao, "in=%d flags=%d space=%d r=%d wa/pl=%d/%d needed=%d more=%d\n",
-             max, flags, space, r, p->wait_on_ao, p->still_playing, needed, more);
-}
-
-static void *playthread(void *arg)
-{
-    struct ao *ao = arg;
-    struct ao_push_state *p = ao->api_priv;
-    mpthread_set_name("ao");
-    pthread_mutex_lock(&p->lock);
-    while (!p->terminate) {
-        bool blocked = ao->driver->initially_blocked && !p->initial_unblocked;
-        bool playing = (!p->paused || ao->stream_silence) && !blocked;
-        if (playing)
-            ao_play_data(ao);
-
-        if (!p->need_wakeup) {
-            MP_STATS(ao, "start audio wait");
-            if (!p->wait_on_ao || !playing) {
-                // Avoid busy waiting, because the audio API will still report
-                // that it needs new data, even if we're not ready yet, or if
-                // get_space() decides that the amount of audio buffered in the
-                // device is enough, and p->buffer can be empty.
-                // The most important part is that the decoder is woken up, so
-                // that the decoder will wake up us in turn.
-                MP_TRACE(ao, "buffer inactive.\n");
-
-                bool was_playing = p->still_playing;
-                double timeout = -1;
-                if (p->still_playing && !p->paused && p->final_chunk &&
-                    !mp_audio_buffer_samples(p->buffer))
-                {
-                    double now = mp_time_sec();
-                    if (!p->expected_end_time)
-                        p->expected_end_time = now + unlocked_get_delay(ao);
-                    if (p->expected_end_time < now) {
-                        p->still_playing = false;
-                    } else {
-                        timeout = p->expected_end_time - now;
-                    }
-                }
-
-                if (was_playing && !p->still_playing)
-                    ao->wakeup_cb(ao->wakeup_ctx);
-                pthread_cond_signal(&p->wakeup); // for draining
-
-                if (p->still_playing && timeout > 0) {
-                    struct timespec ts = mp_rel_time_to_timespec(timeout);
-                    pthread_cond_timedwait(&p->wakeup, &p->lock, &ts);
-                } else {
-                    pthread_cond_wait(&p->wakeup, &p->lock);
-                }
-            } else {
-                // Wait until the device wants us to write more data to it.
-                if (!ao->driver->wait || ao->driver->wait(ao, &p->lock) < 0) {
-                    // Fallback to guessing.
-                    double timeout = 0;
-                    if (ao->driver->get_delay)
-                        timeout = ao->driver->get_delay(ao);
-                    timeout *= 0.25; // wake up if 25% played
-                    if (!p->need_wakeup) {
-                        struct timespec ts = mp_rel_time_to_timespec(timeout);
-                        pthread_cond_timedwait(&p->wakeup, &p->lock, &ts);
-                    }
-                }
-            }
-            MP_STATS(ao, "end audio wait");
-        }
-        p->need_wakeup = false;
-    }
-    pthread_mutex_unlock(&p->lock);
-    return NULL;
-}
-
-static void destroy_no_thread(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-
-    ao->driver->uninit(ao);
-
-    for (int n = 0; n < 2; n++) {
-        int h = p->wakeup_pipe[n];
-        if (h >= 0)
-            close(h);
-    }
-
-    pthread_cond_destroy(&p->wakeup);
-    pthread_mutex_destroy(&p->lock);
-}
-
-static void uninit(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-
-    pthread_mutex_lock(&p->lock);
-    p->terminate = true;
-    wakeup_playthread(ao);
-    pthread_mutex_unlock(&p->lock);
-
-    pthread_join(p->thread, NULL);
-
-    destroy_no_thread(ao);
-}
-
-static int init(struct ao *ao)
-{
-    struct ao_push_state *p = ao->api_priv;
-
-    pthread_mutex_init(&p->lock, NULL);
-    pthread_cond_init(&p->wakeup, NULL);
-    mp_make_wakeup_pipe(p->wakeup_pipe);
-
-    if (ao->device_buffer <= 0) {
-        MP_FATAL(ao, "Couldn't probe device buffer size.\n");
-        goto err;
-    }
-
-    p->buffer = mp_audio_buffer_create(ao);
-    mp_audio_buffer_reinit_fmt(p->buffer, ao->format,
-                               &ao->channels, ao->samplerate);
-    mp_audio_buffer_preallocate_min(p->buffer, ao->buffer);
-    if (pthread_create(&p->thread, NULL, playthread, ao))
-        goto err;
-    return 0;
-err:
-    destroy_no_thread(ao);
-    return -1;
-}
-
-const struct ao_driver ao_api_push = {
-    .init = init,
-    .control = control,
-    .uninit = uninit,
-    .reset = reset,
-    .get_space = get_space,
-    .play = play,
-    .get_delay = get_delay,
-    .pause = audio_pause,
-    .resume = resume,
-    .drain = drain,
-    .get_eof = get_eof,
-    .priv_size = sizeof(struct ao_push_state),
-};
-
-// Must be called locked.
-int ao_play_silence(struct ao *ao, int samples)
-{
-    assert(ao->api == &ao_api_push);
-
-    struct ao_push_state *p = ao->api_priv;
-
-    if (!realloc_silence(ao, samples) || !ao->driver->play)
-        return 0;
-
-    return ao->driver->play(ao, (void **)p->silence, samples, 0);
-}
-
-void ao_unblock(struct ao *ao)
-{
-    if (ao->api == &ao_api_push) {
-        struct ao_push_state *p = ao->api_priv;
-        pthread_mutex_lock(&p->lock);
-        p->need_wakeup = true;
-        p->initial_unblocked = true;
-        wakeup_playthread(ao);
-        pthread_cond_signal(&p->wakeup);
-        pthread_mutex_unlock(&p->lock);
-    }
-}
-
-#ifndef __MINGW32__
-
-#include <poll.h>
-
-#define MAX_POLL_FDS 20
-
-// Call poll() for the given fds. This will extend the given fds with the
-// wakeup pipe, so ao_wakeup_poll() will basically interrupt this function.
-// Unlocks the lock temporarily.
-// Returns <0 on error, 0 on success, 1 if the caller should return immediately.
-int ao_wait_poll(struct ao *ao, struct pollfd *fds, int num_fds,
-                 pthread_mutex_t *lock)
-{
-    struct ao_push_state *p = ao->api_priv;
-    assert(ao->api == &ao_api_push);
-    assert(&p->lock == lock);
-
-    if (num_fds >= MAX_POLL_FDS || p->wakeup_pipe[0] < 0)
-        return -1;
-
-    struct pollfd p_fds[MAX_POLL_FDS];
-    memcpy(p_fds, fds, num_fds * sizeof(p_fds[0]));
-    p_fds[num_fds] = (struct pollfd){
-        .fd = p->wakeup_pipe[0],
-        .events = POLLIN,
-    };
-
-    pthread_mutex_unlock(&p->lock);
-    int r = poll(p_fds, num_fds + 1, -1);
-    r = r < 0 ? -errno : 0;
-    pthread_mutex_lock(&p->lock);
-
-    memcpy(fds, p_fds, num_fds * sizeof(fds[0]));
-    bool wakeup = false;
-    if (p_fds[num_fds].revents & POLLIN) {
-        wakeup = true;
-        // might "drown" some wakeups, but that's ok for our use-case
-        mp_flush_wakeup_pipe(p->wakeup_pipe[0]);
-    }
-    return (r >= 0 || r == -EINTR) ? wakeup : -1;
-}
-
-void ao_wakeup_poll(struct ao *ao)
-{
-    assert(ao->api == &ao_api_push);
-    struct ao_push_state *p = ao->api_priv;
-
-    (void)write(p->wakeup_pipe[1], &(char){0}, 1);
-}
-
-#endif
diff --git a/wscript_build.py b/wscript_build.py
index 34398c10cd..1c7a681a77 100644
--- a/wscript_build.py
+++ b/wscript_build.py
@@ -262,8 +262,7 @@ def build(ctx):
         ( "audio/out/ao_wasapi.c",               "wasapi" ),
         ( "audio/out/ao_wasapi_changenotify.c",  "wasapi" ),
         ( "audio/out/ao_wasapi_utils.c",         "wasapi" ),
-        ( "audio/out/pull.c" ),
-        ( "audio/out/push.c" ),
+        ( "audio/out/buffer.c" ),
 
         ## Core
         ( "common/av_common.c" ),