Adding upstream version 0.38.0.upstream/0.38.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

32 files changed, 1248 insertions, 431 deletions

diff --git a/audio/chmap.c b/audio/chmap.c
index e2b95f4..a56d78d 100644
--- a/audio/chmap.c
+++ b/audio/chmap.c
@@ -239,8 +239,8 @@ void mp_chmap_set_unknown(struct mp_chmap *dst, int num_channels)
     }
 }
 
-// Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
-// Speakers not representable by ffmpeg/libav are dropped.
+// Return the ffmpeg channel layout as in <libavutil/channel_layout.h>.
+// Speakers not representable by ffmpeg are dropped.
 // Warning: this ignores the order of the channels, and will return a channel
 //          mask even if the order is different from libavcodec's.
 //          Also, "unknown" channel maps are translated to non-sense channel
@@ -263,7 +263,7 @@ uint64_t mp_chmap_to_lavc_unchecked(const struct mp_chmap *src)
     return mask;
 }
 
-// Return the ffmpeg/libav channel layout as in <libavutil/channel_layout.h>.
+// Return the ffmpeg channel layout as in <libavutil/channel_layout.h>.
 // Returns 0 if the channel order doesn't match lavc's or if it's invalid.
 uint64_t mp_chmap_to_lavc(const struct mp_chmap *src)
 {
@@ -272,7 +272,7 @@ uint64_t mp_chmap_to_lavc(const struct mp_chmap *src)
     return mp_chmap_to_lavc_unchecked(src);
 }
 
-// Set channel map from the ffmpeg/libav channel layout as in
+// Set channel map from the ffmpeg channel layout as in
 // <libavutil/channel_layout.h>.
 // If the number of channels exceed MP_NUM_CHANNELS, set dst to empty.
 void mp_chmap_from_lavc(struct mp_chmap *dst, uint64_t src)
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index 08b789a..9b5f1fb 100644
--- a/audio/decode/ad_lavc.c
+++ b/audio/decode/ad_lavc.c
@@ -44,6 +44,7 @@
 #include "options/options.h"
 
 struct priv {
+    struct mp_codec_params *codec;
     AVCodecContext *avctx;
     AVFrame *avframe;
     AVPacket *avpkt;
@@ -156,7 +157,7 @@ static bool init(struct mp_filter *da, struct mp_codec_params *codec,
     return true;
 }
 
-static void destroy(struct mp_filter *da)
+static void ad_lavc_destroy(struct mp_filter *da)
 {
     struct priv *ctx = da->priv;
 
@@ -165,7 +166,7 @@ static void destroy(struct mp_filter *da)
     mp_free_av_packet(&ctx->avpkt);
 }
 
-static void reset(struct mp_filter *da)
+static void ad_lavc_reset(struct mp_filter *da)
 {
     struct priv *ctx = da->priv;
 
@@ -219,6 +220,8 @@ static int receive_frame(struct mp_filter *da, struct mp_frame *out)
     if (!priv->avframe->buf[0])
         return ret;
 
+    mp_codec_info_from_av(avctx, priv->codec);
+
     double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
 
     struct mp_aframe *mpframe = mp_aframe_from_avframe(priv->avframe);
@@ -276,7 +279,7 @@ static int receive_frame(struct mp_filter *da, struct mp_frame *out)
     return ret;
 }
 
-static void process(struct mp_filter *ad)
+static void ad_lavc_process(struct mp_filter *ad)
 {
     struct priv *priv = ad->priv;
 
@@ -286,9 +289,9 @@ static void process(struct mp_filter *ad)
 static const struct mp_filter_info ad_lavc_filter = {
     .name = "ad_lavc",
     .priv_size = sizeof(struct priv),
-    .process = process,
-    .reset = reset,
-    .destroy = destroy,
+    .process = ad_lavc_process,
+    .reset = ad_lavc_reset,
+    .destroy = ad_lavc_destroy,
 };
 
 static struct mp_decoder *create(struct mp_filter *parent,
@@ -305,12 +308,16 @@ static struct mp_decoder *create(struct mp_filter *parent,
     da->log = mp_log_new(da, parent->log, NULL);
 
     struct priv *priv = da->priv;
+    priv->codec = codec;
     priv->public.f = da;
 
     if (!init(da, codec, decoder)) {
         talloc_free(da);
         return NULL;
     }
+
+    codec->codec_desc = priv->avctx->codec_descriptor->long_name;
+
     return &priv->public;
 }
 
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
index 393af8a..98a53f3 100644
--- a/audio/decode/ad_spdif.c
+++ b/audio/decode/ad_spdif.c
@@ -37,6 +37,12 @@
 
 #define OUTBUF_SIZE 65536
 
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(60, 26, 100)
+#define AV_PROFILE_UNKNOWN FF_PROFILE_UNKNOWN
+#define AV_PROFILE_DTS_HD_HRA FF_PROFILE_DTS_HD_HRA
+#define AV_PROFILE_DTS_HD_MA FF_PROFILE_DTS_HD_MA
+#endif
+
 struct spdifContext {
     struct mp_log   *log;
     enum AVCodecID   codec_id;
@@ -53,7 +59,11 @@ struct spdifContext {
     struct mp_decoder public;
 };
 
+#if LIBAVCODEC_VERSION_MAJOR < 61
 static int write_packet(void *p, uint8_t *buf, int buf_size)
+#else
+static int write_packet(void *p, const uint8_t *buf, int buf_size)
+#endif
 {
     struct spdifContext *ctx = p;
 
@@ -69,7 +79,7 @@ static int write_packet(void *p, uint8_t *buf, int buf_size)
 }
 
 // (called on both filter destruction _and_ if lavf fails to init)
-static void destroy(struct mp_filter *da)
+static void ad_spdif_destroy(struct mp_filter *da)
 {
     struct spdifContext *spdif_ctx = da->priv;
     AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
@@ -79,7 +89,7 @@ static void destroy(struct mp_filter *da)
             av_write_trailer(lavf_ctx);
         if (lavf_ctx->pb)
             av_freep(&lavf_ctx->pb->buffer);
-        av_freep(&lavf_ctx->pb);
+        avio_context_free(&lavf_ctx->pb);
         avformat_free_context(lavf_ctx);
         spdif_ctx->lavf_ctx = NULL;
     }
@@ -90,7 +100,7 @@ static void determine_codec_params(struct mp_filter *da, AVPacket *pkt,
                                    int *out_profile, int *out_rate)
 {
     struct spdifContext *spdif_ctx = da->priv;
-    int profile = FF_PROFILE_UNKNOWN;
+    int profile = AV_PROFILE_UNKNOWN;
     AVCodecContext *ctx = NULL;
     AVFrame *frame = NULL;
 
@@ -115,7 +125,7 @@ static void determine_codec_params(struct mp_filter *da, AVPacket *pkt,
         av_parser_close(parser);
     }
 
-    if (profile != FF_PROFILE_UNKNOWN || spdif_ctx->codec_id != AV_CODEC_ID_DTS)
+    if (profile != AV_PROFILE_UNKNOWN  || spdif_ctx->codec_id != AV_CODEC_ID_DTS)
         return;
 
     const AVCodec *codec = avcodec_find_decoder(spdif_ctx->codec_id);
@@ -145,7 +155,7 @@ done:
     av_frame_free(&frame);
     avcodec_free_context(&ctx);
 
-    if (profile == FF_PROFILE_UNKNOWN)
+    if (profile == AV_PROFILE_UNKNOWN)
         MP_WARN(da, "Failed to parse codec profile.\n");
 }
 
@@ -155,7 +165,7 @@ static int init_filter(struct mp_filter *da)
 
     AVPacket *pkt = spdif_ctx->avpkt;
 
-    int profile = FF_PROFILE_UNKNOWN;
+    int profile = AV_PROFILE_UNKNOWN;
     int c_rate = 0;
     determine_codec_params(da, pkt, &profile, &c_rate);
     MP_VERBOSE(da, "In: profile=%d samplerate=%d\n", profile, c_rate);
@@ -186,7 +196,8 @@ static int init_filter(struct mp_filter *da)
     if (!stream)
         goto fail;
 
-    stream->codecpar->codec_id = spdif_ctx->codec_id;
+    stream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+    stream->codecpar->codec_id   = spdif_ctx->codec_id;
 
     AVDictionary *format_opts = NULL;
 
@@ -208,15 +219,15 @@ static int init_filter(struct mp_filter *da)
         num_channels                    = 2;
         break;
     case AV_CODEC_ID_DTS: {
-        bool is_hd = profile == FF_PROFILE_DTS_HD_HRA ||
-                     profile == FF_PROFILE_DTS_HD_MA  ||
-                     profile == FF_PROFILE_UNKNOWN;
+        bool is_hd = profile == AV_PROFILE_DTS_HD_HRA ||
+                     profile == AV_PROFILE_DTS_HD_MA  ||
+                     profile == AV_PROFILE_UNKNOWN;
 
         // Apparently, DTS-HD over SPDIF is specified to be 7.1 (8 channels)
         // for DTS-HD MA, and stereo (2 channels) for DTS-HD HRA. The bit
         // streaming rate as well as the signaled channel count are defined
         // based on this value.
-        int dts_hd_spdif_channel_count = profile == FF_PROFILE_DTS_HD_HRA ?
+        int dts_hd_spdif_channel_count = profile == AV_PROFILE_DTS_HD_HRA ?
                                          2 : 8;
         if (spdif_ctx->use_dts_hd && is_hd) {
             av_dict_set_int(&format_opts, "dtshd_rate",
@@ -226,7 +237,7 @@ static int init_filter(struct mp_filter *da)
             num_channels                = dts_hd_spdif_channel_count;
         } else {
             sample_format               = AF_FORMAT_S_DTS;
-            samplerate                  = 48000;
+            samplerate                  = c_rate > 44100 ? 48000 : 44100;
             num_channels                = 2;
         }
         break;
@@ -250,6 +261,8 @@ static int init_filter(struct mp_filter *da)
         abort();
     }
 
+    stream->codecpar->sample_rate = samplerate;
+
     struct mp_chmap chmap;
     mp_chmap_from_channels(&chmap, num_channels);
     mp_aframe_set_chmap(spdif_ctx->fmt, &chmap);
@@ -270,12 +283,12 @@ static int init_filter(struct mp_filter *da)
     return 0;
 
 fail:
-    destroy(da);
+    ad_spdif_destroy(da);
     mp_filter_internal_mark_failed(da);
     return -1;
 }
 
-static void process(struct mp_filter *da)
+static void ad_spdif_process(struct mp_filter *da)
 {
     struct spdifContext *spdif_ctx = da->priv;
 
@@ -400,8 +413,8 @@ struct mp_decoder_list *select_spdif_codec(const char *codec, const char *pref)
 static const struct mp_filter_info ad_spdif_filter = {
     .name = "ad_spdif",
     .priv_size = sizeof(struct spdifContext),
-    .process = process,
-    .destroy = destroy,
+    .process = ad_spdif_process,
+    .destroy = ad_spdif_destroy,
 };
 
 static struct mp_decoder *create(struct mp_filter *parent,
diff --git a/audio/filter/af_drop.c b/audio/filter/af_drop.c
index 724c482..499389d 100644
--- a/audio/filter/af_drop.c
+++ b/audio/filter/af_drop.c
@@ -11,7 +11,7 @@ struct priv {
     struct mp_aframe *last; // for repeating
 };
 
-static void process(struct mp_filter *f)
+static void af_drop_process(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -52,7 +52,7 @@ static void process(struct mp_filter *f)
     mp_pin_in_write(f->ppins[1], frame);
 }
 
-static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
+static bool af_drop_command(struct mp_filter *f, struct mp_filter_command *cmd)
 {
     struct priv *p = f->priv;
 
@@ -65,7 +65,7 @@ static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
     return false;
 }
 
-static void reset(struct mp_filter *f)
+static void af_drop_reset(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -73,18 +73,18 @@ static void reset(struct mp_filter *f)
     p->diff = 0;
 }
 
-static void destroy(struct mp_filter *f)
+static void af_drop_destroy(struct mp_filter *f)
 {
-    reset(f);
+    af_drop_reset(f);
 }
 
 static const struct mp_filter_info af_drop_filter = {
     .name = "drop",
     .priv_size = sizeof(struct priv),
-    .process = process,
-    .command = command,
-    .reset = reset,
-    .destroy = destroy,
+    .process = af_drop_process,
+    .command = af_drop_command,
+    .reset = af_drop_reset,
+    .destroy = af_drop_destroy,
 };
 
 static struct mp_filter *af_drop_create(struct mp_filter *parent, void *options)
diff --git a/audio/filter/af_format.c b/audio/filter/af_format.c
index 2d1c1cc..eddce64 100644
--- a/audio/filter/af_format.c
+++ b/audio/filter/af_format.c
@@ -38,7 +38,7 @@ struct priv {
     struct mp_pin *in_pin;
 };
 
-static void process(struct mp_filter *f)
+static void af_format_process(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -85,7 +85,7 @@ error:
 static const struct mp_filter_info af_format_filter = {
     .name = "format",
     .priv_size = sizeof(struct priv),
-    .process = process,
+    .process = af_format_process,
 };
 
 static struct mp_filter *af_format_create(struct mp_filter *parent,
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index b4a1d59..def9700 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -50,7 +50,7 @@
 #define AC3_MAX_CHANNELS 6
 #define AC3_MAX_CODED_FRAME_SIZE 3840
 #define AC3_FRAME_SIZE (6  * 256)
-const static uint16_t ac3_bitrate_tab[19] = {
+static const uint16_t ac3_bitrate_tab[19] = {
     32, 40, 48, 56, 64, 80, 96, 112, 128,
     160, 192, 224, 256, 320, 384, 448, 512, 576, 640
 };
@@ -103,7 +103,15 @@ static bool reinit(struct mp_filter *f)
     if (!bit_rate && chmap.num < AC3_MAX_CHANNELS + 1)
         bit_rate = default_bit_rate[chmap.num];
 
-    avcodec_close(s->lavc_actx);
+    avcodec_free_context(&s->lavc_actx);
+    s->lavc_actx = avcodec_alloc_context3(s->lavc_acodec);
+    if (!s->lavc_actx) {
+        MP_ERR(f, "Audio LAVC, couldn't reallocate context!\n");
+        return false;
+    }
+
+    if (mp_set_avopts(f->log, s->lavc_actx, s->opts->avopts) < 0)
+        return false;
 
     // Put sample parameters
     s->lavc_actx->sample_fmt = af_to_avformat(format);
@@ -131,18 +139,18 @@ static bool reinit(struct mp_filter *f)
     return true;
 }
 
-static void reset(struct mp_filter *f)
+static void af_lavcac3enc_reset(struct mp_filter *f)
 {
     struct priv *s = f->priv;
 
     TA_FREEP(&s->in_frame);
 }
 
-static void destroy(struct mp_filter *f)
+static void af_lavcac3enc_destroy(struct mp_filter *f)
 {
     struct priv *s = f->priv;
 
-    reset(f);
+    af_lavcac3enc_reset(f);
     av_packet_free(&s->lavc_pkt);
     avcodec_free_context(&s->lavc_actx);
 }
@@ -153,7 +161,7 @@ static void swap_16(uint16_t *ptr, size_t size)
         ptr[n] = av_bswap16(ptr[n]);
 }
 
-static void process(struct mp_filter *f)
+static void af_lavcac3enc_process(struct mp_filter *f)
 {
     struct priv *s = f->priv;
 
@@ -187,9 +195,6 @@ static void process(struct mp_filter *f)
         case MP_FRAME_AUDIO:
             TA_FREEP(&s->in_frame);
             s->in_frame = input.data;
-            frame = mp_frame_to_av(input, NULL);
-            if (!frame)
-                goto error;
             if (mp_aframe_get_channels(s->in_frame) < s->opts->min_channel_num) {
                 // Just pass it through.
                 s->in_frame = NULL;
@@ -200,6 +205,9 @@ static void process(struct mp_filter *f)
                 if (!reinit(f))
                     goto error;
             }
+            frame = mp_frame_to_av(input, NULL);
+            if (!frame)
+                goto error;
             break;
         default: goto error; // unexpected packet type
         }
@@ -273,9 +281,9 @@ error:
 static const struct mp_filter_info af_lavcac3enc_filter = {
     .name = "lavcac3enc",
     .priv_size = sizeof(struct priv),
-    .process = process,
-    .reset = reset,
-    .destroy = destroy,
+    .process = af_lavcac3enc_process,
+    .reset = af_lavcac3enc_reset,
+    .destroy = af_lavcac3enc_destroy,
 };
 
 static void add_chmaps_to_autoconv(struct mp_filter *f,
diff --git a/audio/filter/af_rubberband.c b/audio/filter/af_rubberband.c
index 48e5cc1..e71937f 100644
--- a/audio/filter/af_rubberband.c
+++ b/audio/filter/af_rubberband.c
@@ -105,7 +105,7 @@ static bool init_rubberband(struct mp_filter *f)
     return true;
 }
 
-static void process(struct mp_filter *f)
+static void af_rubberband_process(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -233,7 +233,7 @@ error:
     mp_filter_internal_mark_failed(f);
 }
 
-static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
+static bool af_rubberband_command(struct mp_filter *f, struct mp_filter_command *cmd)
 {
     struct priv *p = f->priv;
 
@@ -263,7 +263,7 @@ static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
     return false;
 }
 
-static void reset(struct mp_filter *f)
+static void af_rubberband_reset(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -274,7 +274,7 @@ static void reset(struct mp_filter *f)
     TA_FREEP(&p->pending);
 }
 
-static void destroy(struct mp_filter *f)
+static void af_rubberband_destroy(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -286,10 +286,10 @@ static void destroy(struct mp_filter *f)
 static const struct mp_filter_info af_rubberband_filter = {
     .name = "rubberband",
     .priv_size = sizeof(struct priv),
-    .process = process,
-    .command = command,
-    .reset = reset,
-    .destroy = destroy,
+    .process = af_rubberband_process,
+    .command = af_rubberband_command,
+    .reset = af_rubberband_reset,
+    .destroy = af_rubberband_destroy,
 };
 
 static struct mp_filter *af_rubberband_create(struct mp_filter *parent,
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index f06478f..e7b101b 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -229,7 +229,7 @@ static void output_overlap_s16(struct priv *s, void *buf_out,
     }
 }
 
-static void process(struct mp_filter *f)
+static void af_scaletempo_process(struct mp_filter *f)
 {
     struct priv *s = f->priv;
 
@@ -511,7 +511,7 @@ static bool reinit(struct mp_filter *f)
     return true;
 }
 
-static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
+static bool af_scaletempo_command(struct mp_filter *f, struct mp_filter_command *cmd)
 {
     struct priv *s = f->priv;
 
@@ -530,7 +530,7 @@ static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
     return false;
 }
 
-static void reset(struct mp_filter *f)
+static void af_scaletempo_reset(struct mp_filter *f)
 {
     struct priv *s = f->priv;
 
@@ -543,7 +543,7 @@ static void reset(struct mp_filter *f)
     TA_FREEP(&s->in);
 }
 
-static void destroy(struct mp_filter *f)
+static void af_scaletempo_destroy(struct mp_filter *f)
 {
     struct priv *s = f->priv;
     free(s->buf_queue);
@@ -558,10 +558,10 @@ static void destroy(struct mp_filter *f)
 static const struct mp_filter_info af_scaletempo_filter = {
     .name = "scaletempo",
     .priv_size = sizeof(struct priv),
-    .process = process,
-    .command = command,
-    .reset = reset,
-    .destroy = destroy,
+    .process = af_scaletempo_process,
+    .command = af_scaletempo_command,
+    .reset = af_scaletempo_reset,
+    .destroy = af_scaletempo_destroy,
 };
 
 static struct mp_filter *af_scaletempo_create(struct mp_filter *parent,
diff --git a/audio/filter/af_scaletempo2.c b/audio/filter/af_scaletempo2.c
index 7ad8e35..e43c29a 100644
--- a/audio/filter/af_scaletempo2.c
+++ b/audio/filter/af_scaletempo2.c
@@ -19,9 +19,9 @@ struct priv {
 };
 
 static bool init_scaletempo2(struct mp_filter *f);
-static void reset(struct mp_filter *f);
+static void af_scaletempo2_reset(struct mp_filter *f);
 
-static void process(struct mp_filter *f)
+static void af_scaletempo2_process(struct mp_filter *f)
 {
     struct priv *p = f->priv;
 
@@ -156,7 +156,7 @@ static bool init_scaletempo2(struct mp_filter *f)
     return true;
 }
 
-static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
+static bool af_scaletempo2_command(struct mp_filter *f, struct mp_filter_command *cmd)
 {
     struct priv *p = f->priv;
 
@@ -169,7 +169,7 @@ static bool command(struct mp_filter *f, struct mp_filter_command *cmd)
     return false;
 }
 
-static void reset(struct mp_filter *f)
+static void af_scaletempo2_reset(struct mp_filter *f)
 {
     struct priv *p = f->priv;
     mp_scaletempo2_reset(&p->data);
@@ -177,7 +177,7 @@ static void reset(struct mp_filter *f)
     TA_FREEP(&p->pending);
 }
 
-static void destroy(struct mp_filter *f)
+static void af_scaletempo2_destroy(struct mp_filter *f)
 {
     struct priv *p = f->priv;
     mp_scaletempo2_destroy(&p->data);
@@ -187,10 +187,10 @@ static void destroy(struct mp_filter *f)
 static const struct mp_filter_info af_scaletempo2_filter = {
     .name = "scaletempo2",
     .priv_size = sizeof(struct priv),
-    .process = process,
-    .command = command,
-    .reset = reset,
-    .destroy = destroy,
+    .process = af_scaletempo2_process,
+    .command = af_scaletempo2_command,
+    .reset = af_scaletempo2_reset,
+    .destroy = af_scaletempo2_destroy,
 };
 
 static struct mp_filter *af_scaletempo2_create(
diff --git a/audio/filter/af_scaletempo2_internals.c b/audio/filter/af_scaletempo2_internals.c
index 534f4f6..924c091 100644
--- a/audio/filter/af_scaletempo2_internals.c
+++ b/audio/filter/af_scaletempo2_internals.c
@@ -93,15 +93,15 @@ static void multi_channel_moving_block_energies(
 }
 
 static float multi_channel_similarity_measure(
-    const float* dot_prod_a_b,
-    const float* energy_a, const float* energy_b,
+    const float* dot_prod,
+    const float* energy_target, const float* energy_candidate,
     int channels)
 {
     const float epsilon = 1e-12f;
     float similarity_measure = 0.0f;
     for (int n = 0; n < channels; ++n) {
-        similarity_measure += dot_prod_a_b[n]
-            / sqrtf(energy_a[n] * energy_b[n] + epsilon);
+        similarity_measure += dot_prod[n] * energy_target[n]
+            / sqrtf(energy_target[n] * energy_candidate[n] + epsilon);
     }
     return similarity_measure;
 }
@@ -765,7 +765,8 @@ double mp_scaletempo2_get_latency(struct mp_scaletempo2 *p, double playback_rate
 
 bool mp_scaletempo2_frames_available(struct mp_scaletempo2 *p, double playback_rate)
 {
-    return p->input_buffer_final_frames > p->target_block_index
+    return (p->input_buffer_final_frames > p->target_block_index &&
+            p->input_buffer_final_frames > 0)
         || can_perform_wsola(p, playback_rate)
         || p->num_complete_frames > 0;
 }
diff --git a/audio/out/ao.c b/audio/out/ao.c
index a5aa3a9..75fcbac 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -40,6 +40,7 @@ extern const struct ao_driver audio_out_audiotrack;
 extern const struct ao_driver audio_out_audiounit;
 extern const struct ao_driver audio_out_coreaudio;
 extern const struct ao_driver audio_out_coreaudio_exclusive;
+extern const struct ao_driver audio_out_avfoundation;
 extern const struct ao_driver audio_out_rsound;
 extern const struct ao_driver audio_out_pipewire;
 extern const struct ao_driver audio_out_sndio;
@@ -65,6 +66,9 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_COREAUDIO
     &audio_out_coreaudio,
 #endif
+#if HAVE_AVFOUNDATION
+    &audio_out_avfoundation,
+#endif
 #if HAVE_PIPEWIRE
     &audio_out_pipewire,
 #endif
@@ -612,7 +616,7 @@ void ao_set_gain(struct ao *ao, float gain)
 
 #define MUL_GAIN_f(d, num_samples, gain)                                        \
     for (int n = 0; n < (num_samples); n++)                                     \
-        (d)[n] = MPCLAMP(((d)[n]) * (gain), -1.0, 1.0)
+        (d)[n] = (d)[n] * (gain)
 
 static void process_plane(struct ao *ao, void *data, int num_samples)
 {
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 75eda3b..92ea0db 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -623,7 +623,8 @@ static void uninit(struct ao *ao)
         CHECK_ALSA_ERROR("pcm close error");
     }
 
-alsa_error: ;
+alsa_error:
+    snd_config_update_free_global();
 }
 
 #define INIT_DEVICE_ERR_GENERIC -1
diff --git a/audio/out/ao_audiotrack.c b/audio/out/ao_audiotrack.c
index 1392699..db1da9c 100644
--- a/audio/out/ao_audiotrack.c
+++ b/audio/out/ao_audiotrack.c
@@ -57,9 +57,6 @@ struct priv {
     bool cfg_pcm_float;
     int cfg_session_id;
 
-    bool needs_timestamp_offset;
-    int64_t timestamp_offset;
-
     bool thread_terminate;
     bool thread_created;
     mp_thread thread;
@@ -67,19 +64,19 @@ struct priv {
     mp_cond wakeup;
 };
 
-struct JNIByteBuffer {
+static struct JNIByteBuffer {
     jclass clazz;
     jmethodID clear;
-    struct MPJniField mapping[];
-} ByteBuffer = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIByteBuffer, member)
-    {"java/nio/ByteBuffer", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
-    {"java/nio/ByteBuffer", "clear", "()Ljava/nio/Buffer;", MP_JNI_METHOD, OFFSET(clear), 1},
+} ByteBuffer;
+#define OFFSET(member) offsetof(struct JNIByteBuffer, member)
+static const struct MPJniField ByteBuffer_mapping[] = {
+    {"java/nio/ByteBuffer", NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
+    {"clear", "()Ljava/nio/Buffer;", MP_JNI_METHOD, OFFSET(clear), 1},
     {0},
-    #undef OFFSET
-}};
+};
+#undef OFFSET
 
-struct JNIAudioTrack {
+static struct JNIAudioTrack {
     jclass clazz;
     jmethodID ctor;
     jmethodID ctorV21;
@@ -110,78 +107,78 @@ struct JNIAudioTrack {
     jint ERROR_INVALID_OPERATION;
     jint WRITE_BLOCKING;
     jint WRITE_NON_BLOCKING;
-    struct MPJniField mapping[];
-} AudioTrack = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioTrack, member)
-    {"android/media/AudioTrack", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
-    {"android/media/AudioTrack", "<init>", "(IIIIIII)V", MP_JNI_METHOD, OFFSET(ctor), 1},
-    {"android/media/AudioTrack", "<init>", "(Landroid/media/AudioAttributes;Landroid/media/AudioFormat;III)V", MP_JNI_METHOD, OFFSET(ctorV21), 0},
-    {"android/media/AudioTrack", "release", "()V", MP_JNI_METHOD, OFFSET(release), 1},
-    {"android/media/AudioTrack", "getState", "()I", MP_JNI_METHOD, OFFSET(getState), 1},
-    {"android/media/AudioTrack", "getPlayState", "()I", MP_JNI_METHOD, OFFSET(getPlayState), 1},
-    {"android/media/AudioTrack", "play", "()V", MP_JNI_METHOD, OFFSET(play), 1},
-    {"android/media/AudioTrack", "stop", "()V", MP_JNI_METHOD, OFFSET(stop), 1},
-    {"android/media/AudioTrack", "flush", "()V", MP_JNI_METHOD, OFFSET(flush), 1},
-    {"android/media/AudioTrack", "pause", "()V", MP_JNI_METHOD, OFFSET(pause), 1},
-    {"android/media/AudioTrack", "write", "([BII)I", MP_JNI_METHOD, OFFSET(write), 1},
-    {"android/media/AudioTrack", "write", "([FIII)I", MP_JNI_METHOD, OFFSET(writeFloat), 1},
-    {"android/media/AudioTrack", "write", "([SIII)I", MP_JNI_METHOD, OFFSET(writeShortV23), 0},
-    {"android/media/AudioTrack", "write", "(Ljava/nio/ByteBuffer;II)I", MP_JNI_METHOD, OFFSET(writeBufferV21), 1},
-    {"android/media/AudioTrack", "getBufferSizeInFrames", "()I", MP_JNI_METHOD, OFFSET(getBufferSizeInFramesV23), 0},
-    {"android/media/AudioTrack", "getTimestamp", "(Landroid/media/AudioTimestamp;)Z", MP_JNI_METHOD, OFFSET(getTimestamp), 1},
-    {"android/media/AudioTrack", "getPlaybackHeadPosition", "()I", MP_JNI_METHOD, OFFSET(getPlaybackHeadPosition), 1},
-    {"android/media/AudioTrack", "getLatency", "()I", MP_JNI_METHOD, OFFSET(getLatency), 1},
-    {"android/media/AudioTrack", "getMinBufferSize", "(III)I", MP_JNI_STATIC_METHOD, OFFSET(getMinBufferSize), 1},
-    {"android/media/AudioTrack", "getNativeOutputSampleRate", "(I)I", MP_JNI_STATIC_METHOD, OFFSET(getNativeOutputSampleRate), 1},
-    {"android/media/AudioTrack", "WRITE_BLOCKING", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(WRITE_BLOCKING), 0},
-    {"android/media/AudioTrack", "WRITE_NON_BLOCKING", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(WRITE_NON_BLOCKING), 0},
-    {"android/media/AudioTrack", "STATE_INITIALIZED", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(STATE_INITIALIZED), 1},
-    {"android/media/AudioTrack", "PLAYSTATE_STOPPED", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(PLAYSTATE_STOPPED), 1},
-    {"android/media/AudioTrack", "PLAYSTATE_PAUSED", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(PLAYSTATE_PAUSED), 1},
-    {"android/media/AudioTrack", "PLAYSTATE_PLAYING", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(PLAYSTATE_PLAYING), 1},
-    {"android/media/AudioTrack", "MODE_STREAM", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(MODE_STREAM), 1},
-    {"android/media/AudioTrack", "ERROR", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR), 1},
-    {"android/media/AudioTrack", "ERROR_BAD_VALUE", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR_BAD_VALUE), 1},
-    {"android/media/AudioTrack", "ERROR_INVALID_OPERATION", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR_INVALID_OPERATION), 1},
+} AudioTrack;
+#define OFFSET(member) offsetof(struct JNIAudioTrack, member)
+static const struct MPJniField AudioTrack_mapping[] = {
+    {"android/media/AudioTrack", NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
+    {"<init>", "(IIIIIII)V", MP_JNI_METHOD, OFFSET(ctor), 1},
+    {"<init>", "(Landroid/media/AudioAttributes;Landroid/media/AudioFormat;III)V", MP_JNI_METHOD, OFFSET(ctorV21), 0},
+    {"release", "()V", MP_JNI_METHOD, OFFSET(release), 1},
+    {"getState", "()I", MP_JNI_METHOD, OFFSET(getState), 1},
+    {"getPlayState", "()I", MP_JNI_METHOD, OFFSET(getPlayState), 1},
+    {"play", "()V", MP_JNI_METHOD, OFFSET(play), 1},
+    {"stop", "()V", MP_JNI_METHOD, OFFSET(stop), 1},
+    {"flush", "()V", MP_JNI_METHOD, OFFSET(flush), 1},
+    {"pause", "()V", MP_JNI_METHOD, OFFSET(pause), 1},
+    {"write", "([BII)I", MP_JNI_METHOD, OFFSET(write), 1},
+    {"write", "([FIII)I", MP_JNI_METHOD, OFFSET(writeFloat), 1},
+    {"write", "([SIII)I", MP_JNI_METHOD, OFFSET(writeShortV23), 0},
+    {"write", "(Ljava/nio/ByteBuffer;II)I", MP_JNI_METHOD, OFFSET(writeBufferV21), 1},
+    {"getBufferSizeInFrames", "()I", MP_JNI_METHOD, OFFSET(getBufferSizeInFramesV23), 0},
+    {"getTimestamp", "(Landroid/media/AudioTimestamp;)Z", MP_JNI_METHOD, OFFSET(getTimestamp), 1},
+    {"getPlaybackHeadPosition", "()I", MP_JNI_METHOD, OFFSET(getPlaybackHeadPosition), 1},
+    {"getLatency", "()I", MP_JNI_METHOD, OFFSET(getLatency), 1},
+    {"getMinBufferSize", "(III)I", MP_JNI_STATIC_METHOD, OFFSET(getMinBufferSize), 1},
+    {"getNativeOutputSampleRate", "(I)I", MP_JNI_STATIC_METHOD, OFFSET(getNativeOutputSampleRate), 1},
+    {"WRITE_BLOCKING", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(WRITE_BLOCKING), 0},
+    {"WRITE_NON_BLOCKING", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(WRITE_NON_BLOCKING), 0},
+    {"STATE_INITIALIZED", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(STATE_INITIALIZED), 1},
+    {"PLAYSTATE_STOPPED", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(PLAYSTATE_STOPPED), 1},
+    {"PLAYSTATE_PAUSED", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(PLAYSTATE_PAUSED), 1},
+    {"PLAYSTATE_PLAYING", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(PLAYSTATE_PLAYING), 1},
+    {"MODE_STREAM", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(MODE_STREAM), 1},
+    {"ERROR", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR), 1},
+    {"ERROR_BAD_VALUE", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR_BAD_VALUE), 1},
+    {"ERROR_INVALID_OPERATION", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR_INVALID_OPERATION), 1},
     {0}
-    #undef OFFSET
-}};
+};
+#undef OFFSET
 
-struct JNIAudioAttributes {
+static struct JNIAudioAttributes {
     jclass clazz;
     jint CONTENT_TYPE_MOVIE;
     jint CONTENT_TYPE_MUSIC;
     jint USAGE_MEDIA;
-    struct MPJniField mapping[];
-} AudioAttributes = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioAttributes, member)
-    {"android/media/AudioAttributes", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 0},
-    {"android/media/AudioAttributes", "CONTENT_TYPE_MOVIE", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CONTENT_TYPE_MOVIE), 0},
-    {"android/media/AudioAttributes", "CONTENT_TYPE_MUSIC", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CONTENT_TYPE_MUSIC), 0},
-    {"android/media/AudioAttributes", "USAGE_MEDIA", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(USAGE_MEDIA), 0},
+} AudioAttributes;
+#define OFFSET(member) offsetof(struct JNIAudioAttributes, member)
+static const struct MPJniField AudioAttributes_mapping[] = {
+    {"android/media/AudioAttributes", NULL, MP_JNI_CLASS, OFFSET(clazz), 0},
+    {"CONTENT_TYPE_MOVIE", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CONTENT_TYPE_MOVIE), 0},
+    {"CONTENT_TYPE_MUSIC", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CONTENT_TYPE_MUSIC), 0},
+    {"USAGE_MEDIA", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(USAGE_MEDIA), 0},
     {0}
-    #undef OFFSET
-}};
+};
+#undef OFFSET
 
-struct JNIAudioAttributesBuilder {
+static struct JNIAudioAttributesBuilder {
     jclass clazz;
     jmethodID ctor;
     jmethodID setUsage;
     jmethodID setContentType;
     jmethodID build;
-    struct MPJniField mapping[];
-} AudioAttributesBuilder = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioAttributesBuilder, member)
-    {"android/media/AudioAttributes$Builder", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 0},
-    {"android/media/AudioAttributes$Builder", "<init>", "()V", MP_JNI_METHOD, OFFSET(ctor), 0},
-    {"android/media/AudioAttributes$Builder", "setUsage", "(I)Landroid/media/AudioAttributes$Builder;", MP_JNI_METHOD, OFFSET(setUsage), 0},
-    {"android/media/AudioAttributes$Builder", "setContentType", "(I)Landroid/media/AudioAttributes$Builder;", MP_JNI_METHOD, OFFSET(setContentType), 0},
-    {"android/media/AudioAttributes$Builder", "build", "()Landroid/media/AudioAttributes;", MP_JNI_METHOD, OFFSET(build), 0},
+} AudioAttributesBuilder;
+#define OFFSET(member) offsetof(struct JNIAudioAttributesBuilder, member)
+static const struct MPJniField AudioAttributesBuilder_mapping[] = {
+    {"android/media/AudioAttributes$Builder", NULL, MP_JNI_CLASS, OFFSET(clazz), 0},
+    {"<init>", "()V", MP_JNI_METHOD, OFFSET(ctor), 0},
+    {"setUsage", "(I)Landroid/media/AudioAttributes$Builder;", MP_JNI_METHOD, OFFSET(setUsage), 0},
+    {"setContentType", "(I)Landroid/media/AudioAttributes$Builder;", MP_JNI_METHOD, OFFSET(setContentType), 0},
+    {"build", "()Landroid/media/AudioAttributes;", MP_JNI_METHOD, OFFSET(build), 0},
     {0}
-    #undef OFFSET
-}};
+};
+#undef OFFSET
 
-struct JNIAudioFormat {
+static struct JNIAudioFormat {
     jclass clazz;
     jint ENCODING_PCM_8BIT;
     jint ENCODING_PCM_16BIT;
@@ -194,77 +191,90 @@ struct JNIAudioFormat {
     jint CHANNEL_OUT_5POINT1;
     jint CHANNEL_OUT_BACK_CENTER;
     jint CHANNEL_OUT_7POINT1_SURROUND;
-    struct MPJniField mapping[];
-} AudioFormat = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioFormat, member)
-    {"android/media/AudioFormat", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
-    {"android/media/AudioFormat", "ENCODING_PCM_8BIT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_PCM_8BIT), 1},
-    {"android/media/AudioFormat", "ENCODING_PCM_16BIT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_PCM_16BIT), 1},
-    {"android/media/AudioFormat", "ENCODING_PCM_FLOAT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_PCM_FLOAT), 1},
-    {"android/media/AudioFormat", "ENCODING_IEC61937", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_IEC61937), 0},
-    {"android/media/AudioFormat", "CHANNEL_OUT_MONO", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_MONO), 1},
-    {"android/media/AudioFormat", "CHANNEL_OUT_STEREO", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_STEREO), 1},
-    {"android/media/AudioFormat", "CHANNEL_OUT_FRONT_CENTER", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_FRONT_CENTER), 1},
-    {"android/media/AudioFormat", "CHANNEL_OUT_QUAD", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_QUAD), 1},
-    {"android/media/AudioFormat", "CHANNEL_OUT_5POINT1", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_5POINT1), 1},
-    {"android/media/AudioFormat", "CHANNEL_OUT_BACK_CENTER", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_BACK_CENTER), 1},
-    {"android/media/AudioFormat", "CHANNEL_OUT_7POINT1_SURROUND", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_7POINT1_SURROUND), 0},
+} AudioFormat;
+#define OFFSET(member) offsetof(struct JNIAudioFormat, member)
+static const struct MPJniField AudioFormat_mapping[] = {
+    {"android/media/AudioFormat", NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
+    {"ENCODING_PCM_8BIT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_PCM_8BIT), 1},
+    {"ENCODING_PCM_16BIT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_PCM_16BIT), 1},
+    {"ENCODING_PCM_FLOAT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_PCM_FLOAT), 1},
+    {"ENCODING_IEC61937", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ENCODING_IEC61937), 0},
+    {"CHANNEL_OUT_MONO", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_MONO), 1},
+    {"CHANNEL_OUT_STEREO", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_STEREO), 1},
+    {"CHANNEL_OUT_FRONT_CENTER", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_FRONT_CENTER), 1},
+    {"CHANNEL_OUT_QUAD", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_QUAD), 1},
+    {"CHANNEL_OUT_5POINT1", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_5POINT1), 1},
+    {"CHANNEL_OUT_BACK_CENTER", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_BACK_CENTER), 1},
+    {"CHANNEL_OUT_7POINT1_SURROUND", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(CHANNEL_OUT_7POINT1_SURROUND), 0},
     {0}
-    #undef OFFSET
-}};
+};
+#undef OFFSET
 
-struct JNIAudioFormatBuilder {
+static struct JNIAudioFormatBuilder {
     jclass clazz;
     jmethodID ctor;
     jmethodID setEncoding;
     jmethodID setSampleRate;
     jmethodID setChannelMask;
     jmethodID build;
-    struct MPJniField mapping[];
-} AudioFormatBuilder = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioFormatBuilder, member)
-    {"android/media/AudioFormat$Builder", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 0},
-    {"android/media/AudioFormat$Builder", "<init>", "()V", MP_JNI_METHOD, OFFSET(ctor), 0},
-    {"android/media/AudioFormat$Builder", "setEncoding", "(I)Landroid/media/AudioFormat$Builder;", MP_JNI_METHOD, OFFSET(setEncoding), 0},
-    {"android/media/AudioFormat$Builder", "setSampleRate", "(I)Landroid/media/AudioFormat$Builder;", MP_JNI_METHOD, OFFSET(setSampleRate), 0},
-    {"android/media/AudioFormat$Builder", "setChannelMask", "(I)Landroid/media/AudioFormat$Builder;", MP_JNI_METHOD, OFFSET(setChannelMask), 0},
-    {"android/media/AudioFormat$Builder", "build", "()Landroid/media/AudioFormat;", MP_JNI_METHOD, OFFSET(build), 0},
+} AudioFormatBuilder;
+#define OFFSET(member) offsetof(struct JNIAudioFormatBuilder, member)
+static const struct MPJniField AudioFormatBuilder_mapping[] = {
+    {"android/media/AudioFormat$Builder", NULL, MP_JNI_CLASS, OFFSET(clazz), 0},
+    {"<init>", "()V", MP_JNI_METHOD, OFFSET(ctor), 0},
+    {"setEncoding", "(I)Landroid/media/AudioFormat$Builder;", MP_JNI_METHOD, OFFSET(setEncoding), 0},
+    {"setSampleRate", "(I)Landroid/media/AudioFormat$Builder;", MP_JNI_METHOD, OFFSET(setSampleRate), 0},
+    {"setChannelMask", "(I)Landroid/media/AudioFormat$Builder;", MP_JNI_METHOD, OFFSET(setChannelMask), 0},
+    {"build", "()Landroid/media/AudioFormat;", MP_JNI_METHOD, OFFSET(build), 0},
     {0}
-    #undef OFFSET
-}};
-
+};
+#undef OFFSET
 
-struct JNIAudioManager {
+static struct JNIAudioManager {
     jclass clazz;
     jint ERROR_DEAD_OBJECT;
     jint STREAM_MUSIC;
-    struct MPJniField mapping[];
-} AudioManager = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioManager, member)
-    {"android/media/AudioManager", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
-    {"android/media/AudioManager", "STREAM_MUSIC", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(STREAM_MUSIC), 1},
-    {"android/media/AudioManager", "ERROR_DEAD_OBJECT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR_DEAD_OBJECT), 0},
+} AudioManager;
+#define OFFSET(member) offsetof(struct JNIAudioManager, member)
+static const struct MPJniField AudioManager_mapping[] = {
+    {"android/media/AudioManager", NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
+    {"STREAM_MUSIC", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(STREAM_MUSIC), 1},
+    {"ERROR_DEAD_OBJECT", "I", MP_JNI_STATIC_FIELD_AS_INT, OFFSET(ERROR_DEAD_OBJECT), 0},
     {0}
-    #undef OFFSET
-}};
+};
+#undef OFFSET
 
-struct JNIAudioTimestamp {
+static struct JNIAudioTimestamp {
     jclass clazz;
     jmethodID ctor;
     jfieldID framePosition;
     jfieldID nanoTime;
-    struct MPJniField mapping[];
-} AudioTimestamp = {.mapping = {
-    #define OFFSET(member) offsetof(struct JNIAudioTimestamp, member)
-    {"android/media/AudioTimestamp", NULL, NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
-    {"android/media/AudioTimestamp", "<init>", "()V", MP_JNI_METHOD, OFFSET(ctor), 1},
-    {"android/media/AudioTimestamp", "framePosition", "J", MP_JNI_FIELD, OFFSET(framePosition), 1},
-    {"android/media/AudioTimestamp", "nanoTime", "J", MP_JNI_FIELD, OFFSET(nanoTime), 1},
+} AudioTimestamp;
+#define OFFSET(member) offsetof(struct JNIAudioTimestamp, member)
+static const struct MPJniField AudioTimestamp_mapping[] = {
+    {"android/media/AudioTimestamp", NULL, MP_JNI_CLASS, OFFSET(clazz), 1},
+    {"<init>", "()V", MP_JNI_METHOD, OFFSET(ctor), 1},
+    {"framePosition", "J", MP_JNI_FIELD, OFFSET(framePosition), 1},
+    {"nanoTime", "J", MP_JNI_FIELD, OFFSET(nanoTime), 1},
     {0}
-    #undef OFFSET
-}};
-
-#define MP_JNI_DELETELOCAL(o) (*env)->DeleteLocalRef(env, o)
+};
+#undef OFFSET
+
+#define ENTRY(name) { &name, name ## _mapping }
+static const struct {
+    void *fields;
+    const struct MPJniField *mapping;
+} jclass_list[] = {
+    ENTRY(ByteBuffer),
+    ENTRY(AudioTrack),
+    ENTRY(AudioAttributes),
+    ENTRY(AudioAttributesBuilder),
+    ENTRY(AudioFormat),
+    ENTRY(AudioFormatBuilder),
+    ENTRY(AudioManager),
+    ENTRY(AudioTimestamp),
+};
+#undef ENTRY
 
 static int AudioTrack_New(struct ao *ao)
 {
@@ -279,24 +289,24 @@ static int AudioTrack_New(struct ao *ao)
         jobject format_builder = MP_JNI_NEW(AudioFormatBuilder.clazz, AudioFormatBuilder.ctor);
         MP_JNI_EXCEPTION_LOG(ao);
         tmp = MP_JNI_CALL_OBJECT(format_builder, AudioFormatBuilder.setEncoding, p->format);
-        MP_JNI_DELETELOCAL(tmp);
+        MP_JNI_LOCAL_FREEP(&tmp);
         tmp = MP_JNI_CALL_OBJECT(format_builder, AudioFormatBuilder.setSampleRate, p->samplerate);
-        MP_JNI_DELETELOCAL(tmp);
+        MP_JNI_LOCAL_FREEP(&tmp);
         tmp = MP_JNI_CALL_OBJECT(format_builder, AudioFormatBuilder.setChannelMask, p->channel_config);
-        MP_JNI_DELETELOCAL(tmp);
+        MP_JNI_LOCAL_FREEP(&tmp);
         jobject format = MP_JNI_CALL_OBJECT(format_builder, AudioFormatBuilder.build);
-        MP_JNI_DELETELOCAL(format_builder);
+        MP_JNI_LOCAL_FREEP(&format_builder);
 
         jobject attr_builder = MP_JNI_NEW(AudioAttributesBuilder.clazz, AudioAttributesBuilder.ctor);
         MP_JNI_EXCEPTION_LOG(ao);
         tmp = MP_JNI_CALL_OBJECT(attr_builder, AudioAttributesBuilder.setUsage, AudioAttributes.USAGE_MEDIA);
-        MP_JNI_DELETELOCAL(tmp);
+        MP_JNI_LOCAL_FREEP(&tmp);
         jint content_type = (ao->init_flags & AO_INIT_MEDIA_ROLE_MUSIC) ?
             AudioAttributes.CONTENT_TYPE_MUSIC : AudioAttributes.CONTENT_TYPE_MOVIE;
         tmp = MP_JNI_CALL_OBJECT(attr_builder, AudioAttributesBuilder.setContentType, content_type);
-        MP_JNI_DELETELOCAL(tmp);
+        MP_JNI_LOCAL_FREEP(&tmp);
         jobject attr = MP_JNI_CALL_OBJECT(attr_builder, AudioAttributesBuilder.build);
-        MP_JNI_DELETELOCAL(attr_builder);
+        MP_JNI_LOCAL_FREEP(&attr_builder);
 
         audiotrack = MP_JNI_NEW(
             AudioTrack.clazz,
@@ -308,8 +318,8 @@ static int AudioTrack_New(struct ao *ao)
             p->cfg_session_id
         );
 
-        MP_JNI_DELETELOCAL(format);
-        MP_JNI_DELETELOCAL(attr);
+        MP_JNI_LOCAL_FREEP(&format);
+        MP_JNI_LOCAL_FREEP(&attr);
     } else {
         MP_VERBOSE(ao, "Using legacy initializer\n");
         audiotrack = MP_JNI_NEW(
@@ -332,7 +342,7 @@ static int AudioTrack_New(struct ao *ao)
     if (MP_JNI_CALL_INT(audiotrack, AudioTrack.getState) != AudioTrack.STATE_INITIALIZED) {
         MP_JNI_CALL_VOID(audiotrack, AudioTrack.release);
         MP_JNI_EXCEPTION_LOG(ao);
-        (*env)->DeleteLocalRef(env, audiotrack);
+        MP_JNI_LOCAL_FREEP(&audiotrack);
         MP_ERR(ao, "AudioTrack.getState failed\n");
         return -1;
     }
@@ -346,7 +356,7 @@ static int AudioTrack_New(struct ao *ao)
     }
 
     p->audiotrack = (*env)->NewGlobalRef(env, audiotrack);
-    (*env)->DeleteLocalRef(env, audiotrack);
+    MP_JNI_LOCAL_FREEP(&audiotrack);
     if (!p->audiotrack)
         return -1;
 
@@ -360,8 +370,7 @@ static int AudioTrack_Recreate(struct ao *ao)
 
     MP_JNI_CALL_VOID(p->audiotrack, AudioTrack.release);
     MP_JNI_EXCEPTION_LOG(ao);
-    (*env)->DeleteGlobalRef(env, p->audiotrack);
-    p->audiotrack = NULL;
+    MP_JNI_GLOBAL_FREEP(&p->audiotrack);
     return AudioTrack_New(ao);
 }
 
@@ -407,11 +416,6 @@ static uint32_t AudioTrack_getPlaybackHeadPosition(struct ao *ao)
         int64_t time = MP_JNI_GET_LONG(p->timestamp, AudioTimestamp.nanoTime);
         if (time == 0)
             fpos = pos = 0;
-        if (p->needs_timestamp_offset) {
-            if (time != 0 && !p->timestamp_offset)
-                p->timestamp_offset = now - time;
-            time += p->timestamp_offset;
-        }
         if (fpos != 0 && time != 0 && state == AudioTrack.PLAYSTATE_PLAYING) {
             double diff = (double)(now - time) / 1e9;
             pos += diff * ao->samplerate;
@@ -497,7 +501,7 @@ static int AudioTrack_write(struct ao *ao, int len)
         // reset positions for reading
         jobject bbuf = MP_JNI_CALL_OBJECT(p->bbuf, ByteBuffer.clear);
         if (MP_JNI_EXCEPTION_LOG(ao) < 0) return -1;
-        (*env)->DeleteLocalRef(env, bbuf);
+        MP_JNI_LOCAL_FREEP(&bbuf);
         ret = MP_JNI_CALL_INT(p->audiotrack, AudioTrack.writeBufferV21, p->bbuf, len, AudioTrack.WRITE_BLOCKING);
         if (MP_JNI_EXCEPTION_LOG(ao) < 0) return -1;
 
@@ -521,35 +525,29 @@ static int AudioTrack_write(struct ao *ao, int len)
 static void uninit_jni(struct ao *ao)
 {
     JNIEnv *env = MP_JNI_GET_ENV(ao);
-    mp_jni_reset_jfields(env, &AudioTrack, AudioTrack.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &AudioTimestamp, AudioTimestamp.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &AudioManager, AudioManager.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &AudioFormat, AudioFormat.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &AudioFormatBuilder, AudioFormatBuilder.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &AudioAttributes, AudioAttributes.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &AudioAttributesBuilder, AudioAttributesBuilder.mapping, 1, ao->log);
-    mp_jni_reset_jfields(env, &ByteBuffer, ByteBuffer.mapping, 1, ao->log);
+    for (int i = 0; i < MP_ARRAY_SIZE(jclass_list); i++) {
+        mp_jni_reset_jfields(env, jclass_list[i].fields,
+                             jclass_list[i].mapping, 1, ao->log);
+    }
 }
 
 static int init_jni(struct ao *ao)
 {
     JNIEnv *env = MP_JNI_GET_ENV(ao);
-    if (mp_jni_init_jfields(env, &AudioTrack, AudioTrack.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &ByteBuffer, ByteBuffer.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &AudioTimestamp, AudioTimestamp.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &AudioManager, AudioManager.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &AudioAttributes, AudioAttributes.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &AudioAttributesBuilder, AudioAttributesBuilder.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &AudioFormatBuilder, AudioFormatBuilder.mapping, 1, ao->log) < 0 ||
-        mp_jni_init_jfields(env, &AudioFormat, AudioFormat.mapping, 1, ao->log) < 0) {
-            uninit_jni(ao);
-            return -1;
+    for (int i = 0; i < MP_ARRAY_SIZE(jclass_list); i++) {
+        if (mp_jni_init_jfields(env, jclass_list[i].fields,
+                                jclass_list[i].mapping, 1, ao->log) < 0) {
+            goto error;
+        }
     }
-
     return 0;
+
+error:
+    uninit_jni(ao);
+    return -1;
 }
 
-static MP_THREAD_VOID playthread(void *arg)
+static MP_THREAD_VOID ao_thread(void *arg)
 {
     struct ao *ao = arg;
     struct priv *p = ao->priv;
@@ -608,34 +606,18 @@ static void uninit(struct ao *ao)
     if (p->audiotrack) {
         MP_JNI_CALL_VOID(p->audiotrack, AudioTrack.release);
         MP_JNI_EXCEPTION_LOG(ao);
-        (*env)->DeleteGlobalRef(env, p->audiotrack);
-        p->audiotrack = NULL;
+        MP_JNI_GLOBAL_FREEP(&p->audiotrack);
     }
 
-    if (p->bytearray) {
-        (*env)->DeleteGlobalRef(env, p->bytearray);
-        p->bytearray = NULL;
-    }
+    MP_JNI_GLOBAL_FREEP(&p->bytearray);
 
-    if (p->shortarray) {
-        (*env)->DeleteGlobalRef(env, p->shortarray);
-        p->shortarray = NULL;
-    }
+    MP_JNI_GLOBAL_FREEP(&p->shortarray);
 
-    if (p->floatarray) {
-        (*env)->DeleteGlobalRef(env, p->floatarray);
-        p->floatarray = NULL;
-    }
+    MP_JNI_GLOBAL_FREEP(&p->floatarray);
 
-    if (p->bbuf) {
-        (*env)->DeleteGlobalRef(env, p->bbuf);
-        p->bbuf = NULL;
-    }
+    MP_JNI_GLOBAL_FREEP(&p->bbuf);
 
-    if (p->timestamp) {
-        (*env)->DeleteGlobalRef(env, p->timestamp);
-        p->timestamp = NULL;
-    }
+    MP_JNI_GLOBAL_FREEP(&p->timestamp);
 
     mp_cond_destroy(&p->wakeup);
     mp_mutex_destroy(&p->lock);
@@ -658,6 +640,10 @@ static int init(struct ao *ao)
 
     if (af_fmt_is_spdif(ao->format)) {
         p->format = AudioFormat.ENCODING_IEC61937;
+        if (!p->format || !AudioTrack.writeShortV23) {
+            MP_ERR(ao, "spdif passthrough not supported by API\n");
+            return -1;
+        }
     } else if (ao->format == AF_FORMAT_U8) {
         p->format = AudioFormat.ENCODING_PCM_8BIT;
     } else if (p->cfg_pcm_float && af_fmt_is_float(ao->format)) {
@@ -752,26 +738,26 @@ static int init(struct ao *ao)
         return -1;
     }
     p->timestamp = (*env)->NewGlobalRef(env, timestamp);
-    (*env)->DeleteLocalRef(env, timestamp);
+    MP_JNI_LOCAL_FREEP(&timestamp);
 
     // decide and create buffer of right type
     if (p->format == AudioFormat.ENCODING_IEC61937) {
         jshortArray shortarray = (*env)->NewShortArray(env, p->chunksize / 2);
         p->shortarray = (*env)->NewGlobalRef(env, shortarray);
-        (*env)->DeleteLocalRef(env, shortarray);
+        MP_JNI_LOCAL_FREEP(&shortarray);
     } else if (AudioTrack.writeBufferV21) {
         MP_VERBOSE(ao, "Using NIO ByteBuffer\n");
         jobject bbuf = (*env)->NewDirectByteBuffer(env, p->chunk, p->chunksize);
         p->bbuf = (*env)->NewGlobalRef(env, bbuf);
-        (*env)->DeleteLocalRef(env, bbuf);
+        MP_JNI_LOCAL_FREEP(&bbuf);
     } else if (p->format == AudioFormat.ENCODING_PCM_FLOAT) {
         jfloatArray floatarray = (*env)->NewFloatArray(env, p->chunksize / sizeof(float));
         p->floatarray = (*env)->NewGlobalRef(env, floatarray);
-        (*env)->DeleteLocalRef(env, floatarray);
+        MP_JNI_LOCAL_FREEP(&floatarray);
     } else {
         jbyteArray bytearray = (*env)->NewByteArray(env, p->chunksize);
         p->bytearray = (*env)->NewGlobalRef(env, bytearray);
-        (*env)->DeleteLocalRef(env, bytearray);
+        MP_JNI_LOCAL_FREEP(&bytearray);
     }
 
     /* create AudioTrack object */
@@ -780,7 +766,7 @@ static int init(struct ao *ao)
         goto error;
     }
 
-    if (mp_thread_create(&p->thread, playthread, ao)) {
+    if (mp_thread_create(&p->thread, ao_thread, ao)) {
         MP_ERR(ao, "pthread creation failed\n");
         goto error;
     }
@@ -812,7 +798,6 @@ static void stop(struct ao *ao)
     p->written_frames = 0;
     p->timestamp_fetched = 0;
     p->timestamp_set = false;
-    p->timestamp_offset = 0;
 }
 
 static void start(struct ao *ao)
diff --git a/audio/out/ao_avfoundation.m b/audio/out/ao_avfoundation.m
new file mode 100644
index 0000000..7654916
--- /dev/null
+++ b/audio/out/ao_avfoundation.m
@@ -0,0 +1,372 @@
+/*
+ * 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 "ao.h"
+#include "audio/format.h"
+#include "audio/out/ao_coreaudio_chmap.h"
+#include "audio/out/ao_coreaudio_utils.h"
+#include "common/common.h"
+#include "common/msg.h"
+#include "internal.h"
+#include "osdep/timer.h"
+#include "ta/ta_talloc.h"
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+#import <CoreAudioTypes/CoreAudioTypes.h>
+#import <CoreFoundation/CoreFoundation.h>
+#import <CoreMedia/CoreMedia.h>
+
+
+@interface AVObserver : NSObject {
+    struct ao *ao;
+}
+- (void)handleRestartNotification:(NSNotification*)notification;
+@end
+
+struct priv {
+    AVSampleBufferAudioRenderer *renderer;
+    AVSampleBufferRenderSynchronizer *synchronizer;
+    dispatch_queue_t queue;
+    CMAudioFormatDescriptionRef format_description;
+    AVObserver *observer;
+    int64_t end_time_av;
+};
+
+static int64_t CMTimeGetNanoseconds(CMTime time)
+{
+    time = CMTimeConvertScale(time, 1000000000, kCMTimeRoundingMethod_Default);
+    return time.value;
+}
+
+static CMTime CMTimeFromNanoseconds(int64_t time)
+{
+    return CMTimeMake(time, 1000000000);
+}
+
+static void feed(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    int samplerate = ao->samplerate;
+    int sstride = ao->sstride;
+
+    CMBlockBufferRef block_buffer = NULL;
+    CMSampleBufferRef sample_buffer = NULL;
+    OSStatus err;
+
+    int request_sample_count = samplerate / 10;
+    int buffer_size = request_sample_count * sstride;
+    void *data[] = {CFAllocatorAllocate(NULL, buffer_size, 0)};
+
+    int64_t cur_time_av = CMTimeGetNanoseconds([p->synchronizer currentTime]);
+    int64_t cur_time_mp = mp_time_ns();
+    int64_t end_time_av = MPMAX(p->end_time_av, cur_time_av);
+    int64_t time_delta = CMTimeGetNanoseconds(CMTimeMake(request_sample_count, samplerate));
+    int real_sample_count = ao_read_data_nonblocking(ao, data, request_sample_count, end_time_av - cur_time_av + cur_time_mp + time_delta);
+    if (real_sample_count == 0) {
+        // avoid spinning by blocking the thread
+        mp_sleep_ns(10000000);
+        goto finish;
+    }
+
+    if ((err = CMBlockBufferCreateWithMemoryBlock(
+        NULL,
+        data[0],
+        buffer_size,
+        NULL,
+        NULL,
+        0,
+        real_sample_count * sstride,
+        0,
+        &block_buffer
+    )) != noErr) {
+        MP_FATAL(ao, "failed to create block buffer\n");
+        MP_VERBOSE(ao, "CMBlockBufferCreateWithMemoryBlock returned %d\n", err);
+        goto error;
+    }
+    data[0] = NULL;
+
+    CMSampleTimingInfo sample_timing_into[] = {(CMSampleTimingInfo) {
+        .duration = CMTimeMake(1, samplerate),
+        .presentationTimeStamp = CMTimeFromNanoseconds(end_time_av),
+        .decodeTimeStamp = kCMTimeInvalid
+    }};
+    size_t sample_size_array[] = {sstride};
+    if ((err = CMSampleBufferCreateReady(
+        NULL,
+        block_buffer,
+        p->format_description,
+        real_sample_count,
+        1,
+        sample_timing_into,
+        1,
+        sample_size_array,
+        &sample_buffer
+    )) != noErr) {
+        MP_FATAL(ao, "failed to create sample buffer\n");
+        MP_VERBOSE(ao, "CMSampleBufferCreateReady returned %d\n", err);
+        goto error;
+    }
+
+    [p->renderer enqueueSampleBuffer:sample_buffer];
+
+    time_delta = CMTimeGetNanoseconds(CMTimeMake(real_sample_count, samplerate));
+    p->end_time_av = end_time_av + time_delta;
+
+    goto finish;
+
+error:
+    ao_request_reload(ao);
+finish:
+    if (data[0]) CFAllocatorDeallocate(NULL, data[0]);
+    if (block_buffer) CFRelease(block_buffer);
+    if (sample_buffer) CFRelease(sample_buffer);
+}
+
+static void start(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    p->end_time_av = -1;
+    [p->synchronizer setRate:1];
+    [p->renderer requestMediaDataWhenReadyOnQueue:p->queue usingBlock:^{
+        feed(ao);
+    }];
+}
+
+static void stop(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    dispatch_sync(p->queue, ^{
+        [p->renderer stopRequestingMediaData];
+        [p->renderer flush];
+        [p->synchronizer setRate:0];
+    });
+}
+
+static bool set_pause(struct ao *ao, bool paused)
+{
+    struct priv *p = ao->priv;
+
+    if (paused) {
+        [p->synchronizer setRate:0];
+    } else {
+        [p->synchronizer setRate:1];
+    }
+
+    return true;
+}
+
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct priv *p = ao->priv;
+
+    switch (cmd) {
+    case AOCONTROL_GET_MUTE:
+        *(bool*)arg = [p->renderer isMuted];
+        return CONTROL_OK;
+    case AOCONTROL_GET_VOLUME:
+        *(float*)arg = [p->renderer volume] * 100;
+        return CONTROL_OK;
+    case AOCONTROL_SET_MUTE:
+        [p->renderer setMuted:*(bool*)arg];
+        return CONTROL_OK;
+    case AOCONTROL_SET_VOLUME:
+        [p->renderer setVolume:*(float*)arg / 100];
+        return CONTROL_OK;
+    default:
+        return CONTROL_UNKNOWN;
+    }
+}
+
+@implementation AVObserver
+- (instancetype)initWithAO:(struct ao*)_ao {
+    self = [super init];
+    if (self) {
+        ao = _ao;
+    }
+    return self;
+}
+- (void)handleRestartNotification:(NSNotification*)notification {
+    char *name = cfstr_get_cstr((CFStringRef)notification.name);
+    MP_WARN(ao, "restarting due to system notification; this will cause desync\n");
+    MP_VERBOSE(ao, "notification name: %s\n", name);
+    talloc_free(name);
+    stop(ao);
+    start(ao);
+}
+@end
+
+static int init(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    AudioChannelLayout *layout = NULL;
+
+#if TARGET_OS_IPHONE
+    AVAudioSession *instance = AVAudioSession.sharedInstance;
+    NSInteger maxChannels = instance.maximumOutputNumberOfChannels;
+    NSInteger prefChannels = MIN(maxChannels, ao->channels.num);
+    [instance setCategory:AVAudioSessionCategoryPlayback error:nil];
+    [instance setMode:AVAudioSessionModeMoviePlayback error:nil];
+    [instance setActive:YES error:nil];
+    [instance setPreferredOutputNumberOfChannels:prefChannels error:nil];
+#endif
+
+    if ((p->renderer = [[AVSampleBufferAudioRenderer alloc] init]) == nil) {
+        MP_FATAL(ao, "failed to create audio renderer\n");
+        MP_VERBOSE(ao, "AVSampleBufferAudioRenderer failed to initialize\n");
+        goto error;
+    }
+    if ((p->synchronizer = [[AVSampleBufferRenderSynchronizer alloc] init]) == nil) {
+        MP_FATAL(ao, "failed to create rendering synchronizer\n");
+        MP_VERBOSE(ao, "AVSampleBufferRenderSynchronizer failed to initialize\n");
+        goto error;
+    }
+    if ((p->queue = dispatch_queue_create(
+        "avfoundation event",
+        dispatch_queue_attr_make_with_qos_class(DISPATCH_QUEUE_SERIAL, QOS_CLASS_USER_INTERACTIVE, 0)
+    )) == NULL) {
+        MP_FATAL(ao, "failed to create dispatch queue\n");
+        MP_VERBOSE(ao, "dispatch_queue_create failed\n");
+        goto error;
+    }
+
+    if (ao->device && ao->device[0]) {
+        [p->renderer setAudioOutputDeviceUniqueID:(NSString*)cfstr_from_cstr(ao->device)];
+    }
+
+    [p->synchronizer addRenderer:p->renderer];
+    if (@available(tvOS 14.5, iOS 14.5, macOS 11.3, *)) {
+        [p->synchronizer setDelaysRateChangeUntilHasSufficientMediaData:NO];
+    }
+
+    if (af_fmt_is_spdif(ao->format)) {
+        MP_FATAL(ao, "avfoundation does not support SPDIF\n");
+#if HAVE_COREAUDIO
+        MP_FATAL(ao, "please use coreaudio_exclusive instead\n");
+#endif
+        goto error;
+    }
+
+    // AVSampleBufferAudioRenderer only supports interleaved formats
+    ao->format = af_fmt_from_planar(ao->format);
+    if (af_fmt_is_planar(ao->format)) {
+        MP_FATAL(ao, "planar audio formats are unsupported\n");
+        goto error;
+    }
+
+    AudioStreamBasicDescription asbd;
+    ca_fill_asbd(ao, &asbd);
+    size_t layout_size = sizeof(AudioChannelLayout)
+                         + (ao->channels.num - 1) * sizeof(AudioChannelDescription);
+    layout = talloc_size(ao, layout_size);
+    layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
+    layout->mNumberChannelDescriptions = ao->channels.num;
+    for (int i = 0; i < ao->channels.num; ++i) {
+        AudioChannelDescription *desc = layout->mChannelDescriptions + i;
+        desc->mChannelFlags = kAudioChannelFlags_AllOff;
+        desc->mChannelLabel = mp_speaker_id_to_ca_label(ao->channels.speaker[i]);
+    }
+
+    void *talloc_ctx = talloc_new(NULL);
+    AudioChannelLayout *std_layout = ca_find_standard_layout(talloc_ctx, layout);
+    memmove(layout, std_layout, sizeof(AudioChannelLayout));
+    talloc_free(talloc_ctx);
+    ca_log_layout(ao, MSGL_V, layout);
+
+    OSStatus err;
+    if ((err = CMAudioFormatDescriptionCreate(
+        NULL,
+        &asbd,
+        layout_size,
+        layout,
+        0,
+        NULL,
+        NULL,
+        &p->format_description
+    )) != noErr) {
+        MP_FATAL(ao, "failed to create audio format description\n");
+        MP_VERBOSE(ao, "CMAudioFormatDescriptionCreate returned %d\n", err);
+        goto error;
+    }
+    talloc_free(layout);
+    layout = NULL;
+
+    // AVSampleBufferAudioRenderer read ahead aggressively
+    ao->device_buffer = ao->samplerate * 2;
+
+    p->observer = [[AVObserver alloc] initWithAO:ao];
+    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+    [center addObserver:p->observer selector:@selector(handleRestartNotification:) name:AVSampleBufferAudioRendererOutputConfigurationDidChangeNotification object:p->renderer];
+    [center addObserver:p->observer selector:@selector(handleRestartNotification:) name:AVSampleBufferAudioRendererWasFlushedAutomaticallyNotification object:p->renderer];
+
+    return CONTROL_OK;
+
+error:
+    talloc_free(layout);
+    if (p->renderer) [p->renderer release];
+    if (p->synchronizer) [p->synchronizer release];
+    if (p->queue) dispatch_release(p->queue);
+    if (p->format_description) CFRelease(p->format_description);
+
+#if TARGET_OS_IPHONE
+    [AVAudioSession.sharedInstance setActive:NO
+        withOptions:AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation
+        error:nil
+    ];
+#endif
+
+    return CONTROL_ERROR;
+}
+
+static void uninit(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    stop(ao);
+
+    [p->renderer release];
+    [p->synchronizer release];
+    dispatch_release(p->queue);
+    CFRelease(p->format_description);
+
+    [[NSNotificationCenter defaultCenter] removeObserver:p->observer];
+    [p->observer release];
+
+#if TARGET_OS_IPHONE
+    [AVAudioSession.sharedInstance setActive:NO
+        withOptions:AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation
+        error:nil
+    ];
+#endif
+}
+
+#define OPT_BASE_STRUCT struct priv
+
+const struct ao_driver audio_out_avfoundation = {
+    .description    = "AVFoundation AVSampleBufferAudioRenderer",
+    .name           = "avfoundation",
+    .uninit         = uninit,
+    .init           = init,
+    .control        = control,
+    .reset          = stop,
+    .start          = start,
+    .set_pause      = set_pause,
+    .list_devs      = ca_get_device_list,
+    .priv_size      = sizeof(struct priv),
+};
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index 37f1313..ae743c9 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -27,6 +27,11 @@
 #include "ao_coreaudio_properties.h"
 #include "ao_coreaudio_utils.h"
 
+// The timeout for stopping the audio unit after being reset. This allows the
+// device to sleep after playback paused. The duration is chosen to match the
+// behavior of AVFoundation.
+#define IDLE_TIME 7 * NSEC_PER_SEC
+
 struct priv {
     AudioDeviceID device;
     AudioUnit audio_unit;
@@ -37,6 +42,12 @@ struct priv {
     AudioStreamID original_asbd_stream;
 
     bool change_physical_format;
+
+    // Block that is executed after `IDLE_TIME` to stop audio output unit.
+    dispatch_block_t idle_work;
+    dispatch_queue_t queue;
+
+    int hotplug_cb_registration_times;
 };
 
 static int64_t ca_get_hardware_latency(struct ao *ao) {
@@ -78,7 +89,7 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
 
     int64_t end = mp_time_ns();
     end += p->hw_latency_ns + ca_get_latency(ts) + ca_frames_to_ns(ao, frames);
-    int samples = ao_read_data_nonblocking(ao, planes, frames, end);
+    int samples = ao_read_data(ao, planes, frames, end);
 
     if (samples == 0)
         *aflags |= kAudioUnitRenderAction_OutputIsSilence;
@@ -128,6 +139,9 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 
 static bool init_audiounit(struct ao *ao, AudioStreamBasicDescription asbd);
 static void init_physical_format(struct ao *ao);
+static void reinit_latency(struct ao *ao);
+static bool register_hotplug_cb(struct ao *ao);
+static void unregister_hotplug_cb(struct ao *ao);
 
 static bool reinit_device(struct ao *ao) {
     struct priv *p = ao->priv;
@@ -154,6 +168,9 @@ static int init(struct ao *ao)
     if (!reinit_device(ao))
         goto coreaudio_error;
 
+    if (!register_hotplug_cb(ao))
+        goto coreaudio_error;
+
     if (p->change_physical_format)
         init_physical_format(ao);
 
@@ -166,6 +183,11 @@ static int init(struct ao *ao)
     if (!init_audiounit(ao, asbd))
         goto coreaudio_error;
 
+    reinit_latency(ao);
+
+    p->queue = dispatch_queue_create("io.mpv.coreaudio_stop_during_idle",
+                                     DISPATCH_QUEUE_SERIAL);
+
     return CONTROL_OK;
 
 coreaudio_error:
@@ -295,8 +317,6 @@ static bool init_audiounit(struct ao *ao, AudioStreamBasicDescription asbd)
     CHECK_CA_ERROR_L(coreaudio_error_audiounit,
                      "can't link audio unit to selected device");
 
-    p->hw_latency_ns = ca_get_hardware_latency(ao);
-
     AURenderCallbackStruct render_cb = (AURenderCallbackStruct) {
         .inputProc       = render_cb_lpcm,
         .inputProcRefCon = ao,
@@ -320,24 +340,96 @@ coreaudio_error:
     return false;
 }
 
-static void reset(struct ao *ao)
+static void reinit_latency(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    p->hw_latency_ns = ca_get_hardware_latency(ao);
+}
+
+static void stop(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    OSStatus err = AudioOutputUnitStop(p->audio_unit);
+    CHECK_CA_WARN("can't stop audio unit");
+}
+
+static void cancel_and_release_idle_work(struct priv *p)
+{
+    if (!p->idle_work)
+        return;
+
+    dispatch_block_cancel(p->idle_work);
+    Block_release(p->idle_work);
+    p->idle_work = NULL;
+}
+
+static void stop_after_idle_time(struct ao *ao)
 {
     struct priv *p = ao->priv;
+
+    cancel_and_release_idle_work(p);
+
+    p->idle_work = dispatch_block_create(0, ^{
+        MP_VERBOSE(ao, "Stopping audio unit due to idle timeout\n");
+        stop(ao);
+    });
+
+    dispatch_after(dispatch_time(DISPATCH_TIME_NOW, IDLE_TIME),
+                   p->queue, p->idle_work);
+}
+
+static void _reset(void *_ao)
+{
+    struct ao *ao = (struct ao *)_ao;
+    struct priv *p = ao->priv;
     OSStatus err = AudioUnitReset(p->audio_unit, kAudioUnitScope_Global, 0);
     CHECK_CA_WARN("can't reset audio unit");
+
+    // Until the audio unit is stopped the macOS daemon coreaudiod continues to
+    // consume CPU and prevent macOS from sleeping. Immediately stopping the
+    // audio unit would be disruptive for short pause/resume cycles as
+    // restarting the audio unit takes a noticeable amount of time when a
+    // wireless audio device is being used. Instead the audio unit is stopped
+    // after a delay if it remains idle.
+    stop_after_idle_time(ao);
 }
 
-static void start(struct ao *ao)
+static void reset(struct ao *ao)
 {
     struct priv *p = ao->priv;
+    // Must dispatch to serialize reset, start and stop operations.
+    dispatch_sync_f(p->queue, ao, &_reset);
+}
+
+static void _start(void *_ao)
+{
+    struct ao *ao = (struct ao *)_ao;
+    struct priv *p = ao->priv;
+
+    if (p->idle_work)
+        dispatch_block_cancel(p->idle_work);
+
     OSStatus err = AudioOutputUnitStart(p->audio_unit);
     CHECK_CA_WARN("can't start audio unit");
 }
 
+static void start(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    // Must dispatch to serialize reset, start and stop operations.
+    dispatch_sync_f(p->queue, ao, &_start);
+}
 
 static void uninit(struct ao *ao)
 {
     struct priv *p = ao->priv;
+
+    dispatch_sync(p->queue, ^{
+        cancel_and_release_idle_work(p);
+    });
+    dispatch_release(p->queue);
+
     AudioOutputUnitStop(p->audio_unit);
     AudioUnitUninitialize(p->audio_unit);
     AudioComponentInstanceDispose(p->audio_unit);
@@ -348,6 +440,8 @@ static void uninit(struct ao *ao)
                               &p->original_asbd);
         CHECK_CA_WARN("could not restore physical stream format");
     }
+
+    unregister_hotplug_cb(ao);
 }
 
 static OSStatus hotplug_cb(AudioObjectID id, UInt32 naddr,
@@ -355,8 +449,11 @@ static OSStatus hotplug_cb(AudioObjectID id, UInt32 naddr,
                            void *ctx)
 {
     struct ao *ao = ctx;
+    struct priv *p = ao->priv;
     MP_VERBOSE(ao, "Handling potential hotplug event...\n");
     reinit_device(ao);
+    if (p->audio_unit)
+        reinit_latency(ao);
     ao_hotplug_event(ao);
     return noErr;
 }
@@ -369,7 +466,25 @@ static uint32_t hotplug_properties[] = {
 static int hotplug_init(struct ao *ao)
 {
     if (!reinit_device(ao))
-        goto coreaudio_error;
+        return -1;
+
+    if (!register_hotplug_cb(ao))
+        return -1;
+
+    return 0;
+}
+
+static void hotplug_uninit(struct ao *ao)
+{
+    unregister_hotplug_cb(ao);
+}
+
+static bool register_hotplug_cb(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    if (p->hotplug_cb_registration_times++)
+        return true;
 
     OSStatus err = noErr;
     for (int i = 0; i < MP_ARRAY_SIZE(hotplug_properties); i++) {
@@ -388,14 +503,19 @@ static int hotplug_init(struct ao *ao)
         }
     }
 
-    return 0;
+    return true;
 
 coreaudio_error:
-    return -1;
+    return false;
 }
 
-static void hotplug_uninit(struct ao *ao)
+static void unregister_hotplug_cb(struct ao *ao)
 {
+    struct priv *p = ao->priv;
+
+    if (--p->hotplug_cb_registration_times)
+        return;
+
     OSStatus err = noErr;
     for (int i = 0; i < MP_ARRAY_SIZE(hotplug_properties); i++) {
         AudioObjectPropertyAddress addr = {
diff --git a/audio/out/ao_coreaudio_chmap.c b/audio/out/ao_coreaudio_chmap.c
index 3fd9550..5a129c4 100644
--- a/audio/out/ao_coreaudio_chmap.c
+++ b/audio/out/ao_coreaudio_chmap.c
@@ -22,6 +22,7 @@
 #include "ao_coreaudio_utils.h"
 
 #include "ao_coreaudio_chmap.h"
+#include <CoreAudioTypes/CoreAudioTypes.h>
 
 static const int speaker_map[][2] = {
     { kAudioChannelLabel_Left,                 MP_SPEAKER_ID_FL   },
@@ -65,6 +66,119 @@ static const int speaker_map[][2] = {
     { 0,                                       -1                 },
 };
 
+static const AudioChannelLayoutTag std_layouts[] = {
+    (100U<<16) | 1, // kAudioChannelLayoutTag_Mono
+    (101U<<16) | 2, // kAudioChannelLayoutTag_Stereo
+    (102U<<16) | 2, // kAudioChannelLayoutTag_StereoHeadphones
+    (103U<<16) | 2, // kAudioChannelLayoutTag_MatrixStereo
+    (104U<<16) | 2, // kAudioChannelLayoutTag_MidSide
+    (105U<<16) | 2, // kAudioChannelLayoutTag_XY
+    (106U<<16) | 2, // kAudioChannelLayoutTag_Binaural
+    (107U<<16) | 4, // kAudioChannelLayoutTag_Ambisonic_B_Format
+    (108U<<16) | 4, // kAudioChannelLayoutTag_Quadraphonic
+    (109U<<16) | 5, // kAudioChannelLayoutTag_Pentagonal
+    (110U<<16) | 6, // kAudioChannelLayoutTag_Hexagonal
+    (111U<<16) | 8, // kAudioChannelLayoutTag_Octagonal
+    (112U<<16) | 8, // kAudioChannelLayoutTag_Cube
+    (113U<<16) | 3, // kAudioChannelLayoutTag_MPEG_3_0_A
+    (114U<<16) | 3, // kAudioChannelLayoutTag_MPEG_3_0_B
+    (115U<<16) | 4, // kAudioChannelLayoutTag_MPEG_4_0_A
+    (116U<<16) | 4, // kAudioChannelLayoutTag_MPEG_4_0_B
+    (117U<<16) | 5, // kAudioChannelLayoutTag_MPEG_5_0_A
+    (118U<<16) | 5, // kAudioChannelLayoutTag_MPEG_5_0_B
+    (119U<<16) | 5, // kAudioChannelLayoutTag_MPEG_5_0_C
+    (120U<<16) | 5, // kAudioChannelLayoutTag_MPEG_5_0_D
+    (121U<<16) | 6, // kAudioChannelLayoutTag_MPEG_5_1_A
+    (122U<<16) | 6, // kAudioChannelLayoutTag_MPEG_5_1_B
+    (123U<<16) | 6, // kAudioChannelLayoutTag_MPEG_5_1_C
+    (124U<<16) | 6, // kAudioChannelLayoutTag_MPEG_5_1_D
+    (125U<<16) | 7, // kAudioChannelLayoutTag_MPEG_6_1_A
+    (126U<<16) | 8, // kAudioChannelLayoutTag_MPEG_7_1_A
+    (127U<<16) | 8, // kAudioChannelLayoutTag_MPEG_7_1_B
+    (128U<<16) | 8, // kAudioChannelLayoutTag_MPEG_7_1_C
+    (129U<<16) | 8, // kAudioChannelLayoutTag_Emagic_Default_7_1
+    (130U<<16) | 8, // kAudioChannelLayoutTag_SMPTE_DTV
+    (131U<<16) | 3, // kAudioChannelLayoutTag_ITU_2_1
+    (132U<<16) | 4, // kAudioChannelLayoutTag_ITU_2_2
+    (133U<<16) | 3, // kAudioChannelLayoutTag_DVD_4
+    (134U<<16) | 4, // kAudioChannelLayoutTag_DVD_5
+    (135U<<16) | 5, // kAudioChannelLayoutTag_DVD_6
+    (136U<<16) | 4, // kAudioChannelLayoutTag_DVD_10
+    (137U<<16) | 5, // kAudioChannelLayoutTag_DVD_11
+    (138U<<16) | 5, // kAudioChannelLayoutTag_DVD_18
+    (139U<<16) | 6, // kAudioChannelLayoutTag_AudioUnit_6_0
+    (140U<<16) | 7, // kAudioChannelLayoutTag_AudioUnit_7_0
+    (148U<<16) | 7, // kAudioChannelLayoutTag_AudioUnit_7_0_Front
+    (141U<<16) | 6, // kAudioChannelLayoutTag_AAC_6_0
+    (142U<<16) | 7, // kAudioChannelLayoutTag_AAC_6_1
+    (143U<<16) | 7, // kAudioChannelLayoutTag_AAC_7_0
+    (183U<<16) | 8, // kAudioChannelLayoutTag_AAC_7_1_B
+    (184U<<16) | 8, // kAudioChannelLayoutTag_AAC_7_1_C
+    (144U<<16) | 8, // kAudioChannelLayoutTag_AAC_Octagonal
+    (145U<<16) | 16, // kAudioChannelLayoutTag_TMH_10_2_std
+    (146U<<16) | 21, // kAudioChannelLayoutTag_TMH_10_2_full
+    (149U<<16) | 2, // kAudioChannelLayoutTag_AC3_1_0_1
+    (150U<<16) | 3, // kAudioChannelLayoutTag_AC3_3_0
+    (151U<<16) | 4, // kAudioChannelLayoutTag_AC3_3_1
+    (152U<<16) | 4, // kAudioChannelLayoutTag_AC3_3_0_1
+    (153U<<16) | 4, // kAudioChannelLayoutTag_AC3_2_1_1
+    (154U<<16) | 5, // kAudioChannelLayoutTag_AC3_3_1_1
+    (155U<<16) | 6, // kAudioChannelLayoutTag_EAC_6_0_A
+    (156U<<16) | 7, // kAudioChannelLayoutTag_EAC_7_0_A
+    (157U<<16) | 7, // kAudioChannelLayoutTag_EAC3_6_1_A
+    (158U<<16) | 7, // kAudioChannelLayoutTag_EAC3_6_1_B
+    (159U<<16) | 7, // kAudioChannelLayoutTag_EAC3_6_1_C
+    (160U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_A
+    (161U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_B
+    (162U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_C
+    (163U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_D
+    (164U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_E
+    (165U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_F
+    (166U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_G
+    (167U<<16) | 8, // kAudioChannelLayoutTag_EAC3_7_1_H
+    (168U<<16) | 4, // kAudioChannelLayoutTag_DTS_3_1
+    (169U<<16) | 5, // kAudioChannelLayoutTag_DTS_4_1
+    (170U<<16) | 6, // kAudioChannelLayoutTag_DTS_6_0_A
+    (171U<<16) | 6, // kAudioChannelLayoutTag_DTS_6_0_B
+    (172U<<16) | 6, // kAudioChannelLayoutTag_DTS_6_0_C
+    (173U<<16) | 7, // kAudioChannelLayoutTag_DTS_6_1_A
+    (174U<<16) | 7, // kAudioChannelLayoutTag_DTS_6_1_B
+    (175U<<16) | 7, // kAudioChannelLayoutTag_DTS_6_1_C
+    (176U<<16) | 7, // kAudioChannelLayoutTag_DTS_7_0
+    (177U<<16) | 8, // kAudioChannelLayoutTag_DTS_7_1
+    (178U<<16) | 8, // kAudioChannelLayoutTag_DTS_8_0_A
+    (179U<<16) | 8, // kAudioChannelLayoutTag_DTS_8_0_B
+    (180U<<16) | 9, // kAudioChannelLayoutTag_DTS_8_1_A
+    (181U<<16) | 9, // kAudioChannelLayoutTag_DTS_8_1_B
+    (182U<<16) | 7, // kAudioChannelLayoutTag_DTS_6_1_D
+    (185U<<16) | 4, // kAudioChannelLayoutTag_WAVE_4_0_B
+    (186U<<16) | 5, // kAudioChannelLayoutTag_WAVE_5_0_B
+    (187U<<16) | 6, // kAudioChannelLayoutTag_WAVE_5_1_B
+    (188U<<16) | 7, // kAudioChannelLayoutTag_WAVE_6_1
+    (189U<<16) | 8, // kAudioChannelLayoutTag_WAVE_7_1
+    (194U<<16) | 8, // kAudioChannelLayoutTag_Atmos_5_1_2
+    (195U<<16) | 10, // kAudioChannelLayoutTag_Atmos_5_1_4
+    (196U<<16) | 10, // kAudioChannelLayoutTag_Atmos_7_1_2
+    (192U<<16) | 12, // kAudioChannelLayoutTag_Atmos_7_1_4
+    (193U<<16) | 16, // kAudioChannelLayoutTag_Atmos_9_1_6
+    (197U<<16) | 4, // kAudioChannelLayoutTag_Logic_4_0_C
+    (198U<<16) | 6, // kAudioChannelLayoutTag_Logic_6_0_B
+    (199U<<16) | 7, // kAudioChannelLayoutTag_Logic_6_1_B
+    (200U<<16) | 7, // kAudioChannelLayoutTag_Logic_6_1_D
+    (201U<<16) | 8, // kAudioChannelLayoutTag_Logic_7_1_B
+    (202U<<16) | 12, // kAudioChannelLayoutTag_Logic_Atmos_7_1_4_B
+    (203U<<16) | 14, // kAudioChannelLayoutTag_Logic_Atmos_7_1_6
+    (204U<<16) | 24, // kAudioChannelLayoutTag_CICP_13
+    (205U<<16) | 8, // kAudioChannelLayoutTag_CICP_14
+    (206U<<16) | 12, // kAudioChannelLayoutTag_CICP_15
+    (207U<<16) | 10, // kAudioChannelLayoutTag_CICP_16
+    (208U<<16) | 12, // kAudioChannelLayoutTag_CICP_17
+    (209U<<16) | 14, // kAudioChannelLayoutTag_CICP_18
+    (210U<<16) | 12, // kAudioChannelLayoutTag_CICP_19
+    (211U<<16) | 14, // kAudioChannelLayoutTag_CICP_20
+    kAudioChannelLayoutTag_Unknown
+};
+
 int ca_label_to_mp_speaker_id(AudioChannelLabel label)
 {
     for (int i = 0; speaker_map[i][1] >= 0; i++)
@@ -73,30 +187,48 @@ int ca_label_to_mp_speaker_id(AudioChannelLabel label)
     return -1;
 }
 
+AudioChannelLabel mp_speaker_id_to_ca_label(int speaker_id)
+{
+    for (int i = 0; speaker_map[i][1] >= 0; i++)
+        if (speaker_map[i][1] == speaker_id)
+            return speaker_map[i][0];
+    return -1; // kAudioChannelLabel_Unknown
+}
+
 #if HAVE_COREAUDIO
-static void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout)
+void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout)
 {
     if (!mp_msg_test(ao->log, l))
         return;
 
-    AudioChannelDescription *descs = layout->mChannelDescriptions;
-
-    mp_msg(ao->log, l, "layout: tag: <%u>, bitmap: <%u>, "
-                       "descriptions <%u>\n",
-                       (unsigned) layout->mChannelLayoutTag,
-                       (unsigned) layout->mChannelBitmap,
-                       (unsigned) layout->mNumberChannelDescriptions);
-
-    for (int i = 0; i < layout->mNumberChannelDescriptions; i++) {
-        AudioChannelDescription d = descs[i];
-        mp_msg(ao->log, l, " - description %d: label <%u, %u>, "
-            " flags: <%u>, coords: <%f, %f, %f>\n", i,
-            (unsigned) d.mChannelLabel,
-            (unsigned) ca_label_to_mp_speaker_id(d.mChannelLabel),
-            (unsigned) d.mChannelFlags,
-            d.mCoordinates[0],
-            d.mCoordinates[1],
-            d.mCoordinates[2]);
+    AudioChannelLayoutTag tag = layout->mChannelLayoutTag;
+    mp_msg(ao->log, l, "audio channel layout: tag: <%u>", tag);
+
+    if (tag == kAudioChannelLayoutTag_UseChannelDescriptions) {
+        AudioChannelDescription *descs = layout->mChannelDescriptions;
+        mp_msg(ao->log, l, ", descriptions <%u>\n",
+                           (unsigned) layout->mNumberChannelDescriptions);
+
+        for (int i = 0; i < layout->mNumberChannelDescriptions; i++) {
+            AudioChannelDescription d = descs[i];
+            mp_msg(ao->log, l, " - description %d: label <%u, %u>,  flags: <%u>",
+                i,
+                (unsigned) d.mChannelLabel,
+                (unsigned) ca_label_to_mp_speaker_id(d.mChannelLabel),
+                (unsigned) d.mChannelFlags);
+            if (d.mChannelFlags != kAudioChannelFlags_AllOff) {
+                mp_msg(ao->log, l, ", coords: <%f, %f, %f>\n",
+                    d.mCoordinates[0],
+                    d.mCoordinates[1],
+                    d.mCoordinates[2]);
+            } else {
+                mp_msg(ao->log, l, "\n");
+            }
+        }
+    } else if (tag == kAudioChannelLayoutTag_UseChannelBitmap) {
+        mp_msg(ao->log, l, ", bitmap <%u>\n", layout->mChannelBitmap);
+    } else {
+        mp_msg(ao->log, l, "\n");
     }
 }
 
@@ -117,7 +249,7 @@ static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
             kAudioFormatProperty_ChannelLayoutForBitmap,
             sizeof(uint32_t), &l->mChannelBitmap, &psize);
         CHECK_CA_ERROR("failed to convert channel bitmap to descriptions (info)");
-        r = talloc_size(NULL, psize);
+        r = talloc_size(talloc_ctx, psize);
         err = AudioFormatGetProperty(
             kAudioFormatProperty_ChannelLayoutForBitmap,
             sizeof(uint32_t), &l->mChannelBitmap, &psize, r);
@@ -127,7 +259,7 @@ static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
         err = AudioFormatGetPropertyInfo(
             kAudioFormatProperty_ChannelLayoutForTag,
             sizeof(AudioChannelLayoutTag), &l->mChannelLayoutTag, &psize);
-        r = talloc_size(NULL, psize);
+        r = talloc_size(talloc_ctx, psize);
         CHECK_CA_ERROR("failed to convert channel tag to descriptions (info)");
         err = AudioFormatGetProperty(
             kAudioFormatProperty_ChannelLayoutForTag,
@@ -135,14 +267,53 @@ static AudioChannelLayout *ca_layout_to_custom_layout(struct ao *ao,
         CHECK_CA_ERROR("failed to convert channel tag to descriptions (get)");
     }
 
-    MP_VERBOSE(ao, "converted input channel layout:\n");
-    ca_log_layout(ao, MSGL_V, l);
+    if (ao) {
+        MP_VERBOSE(ao, "converted input channel layout:\n");
+        ca_log_layout(ao, MSGL_V, l);
+    }
 
     return r;
 coreaudio_error:
     return NULL;
 }
 
+AudioChannelLayout *ca_find_standard_layout(void *talloc_ctx, AudioChannelLayout *l)
+{
+    if (l->mChannelLayoutTag != kAudioChannelLayoutTag_UseChannelDescriptions)
+        return l;
+
+    AudioChannelLayout *s = talloc_size(talloc_ctx, sizeof(AudioChannelLayout));
+
+    for (int i = 0; ; ++i) {
+        if ((s->mChannelLayoutTag = std_layouts[i]) == kAudioChannelLayoutTag_Unknown) {
+            s = NULL;
+            break;
+        }
+
+        AudioChannelLayout *r = ca_layout_to_custom_layout(NULL, talloc_ctx, s);
+
+        if (!r)
+            goto mismatch;
+        if (l->mNumberChannelDescriptions != r->mNumberChannelDescriptions)
+            goto mismatch;
+
+        for (int i = 0; i < l->mNumberChannelDescriptions; ++i) {
+            AudioChannelDescription *ld = l->mChannelDescriptions + i;
+            AudioChannelDescription *rd = r->mChannelDescriptions + i;
+            if (ld->mChannelLabel == rd->mChannelLabel)
+                continue;
+            // XXX: we cannot handle channels with coordinates
+            goto mismatch;
+        }
+
+        break;
+
+mismatch:;
+    }
+
+    return s ? s : l;
+}
+
 
 #define CHMAP(n, ...) &(struct mp_chmap) MP_CONCAT(MP_CHMAP, n) (__VA_ARGS__)
 
@@ -241,8 +412,8 @@ static AudioChannelLayout* ca_query_stereo_layout(struct ao *ao,
                                                   void *talloc_ctx)
 {
     OSStatus err;
-    const int nch = 2;
-    uint32_t channels[nch];
+    uint32_t channels[2];
+    const int nch = MP_ARRAY_SIZE(channels);
     AudioChannelLayout *r = NULL;
 
     AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
diff --git a/audio/out/ao_coreaudio_chmap.h b/audio/out/ao_coreaudio_chmap.h
index b6d160c..0b21e83 100644
--- a/audio/out/ao_coreaudio_chmap.h
+++ b/audio/out/ao_coreaudio_chmap.h
@@ -18,15 +18,22 @@
 #ifndef MPV_COREAUDIO_CHMAP_H
 #define MPV_COREAUDIO_CHMAP_H
 
+#include "config.h"
 #include <AudioToolbox/AudioToolbox.h>
 
-#include "config.h"
+#if HAVE_AVFOUNDATION || HAVE_AUDIOUNIT
+#undef HAVE_COREAUDIO
+#define HAVE_COREAUDIO 1
+#endif
 
 struct mp_chmap;
 
 int ca_label_to_mp_speaker_id(AudioChannelLabel label);
+AudioChannelLabel mp_speaker_id_to_ca_label(int speaker_id);
 
 #if HAVE_COREAUDIO
+AudioChannelLayout *ca_find_standard_layout(void *talloc_ctx, AudioChannelLayout *l);
+void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout);
 bool ca_init_chmap(struct ao *ao, AudioDeviceID device);
 void ca_get_active_chmap(struct ao *ao, AudioDeviceID device, int channel_count,
                          struct mp_chmap *out_map);
diff --git a/audio/out/ao_coreaudio_exclusive.c b/audio/out/ao_coreaudio_exclusive.c
index e24f791..5e0ec3b 100644
--- a/audio/out/ao_coreaudio_exclusive.c
+++ b/audio/out/ao_coreaudio_exclusive.c
@@ -1,5 +1,5 @@
 /*
- * CoreAudio audio output driver for Mac OS X
+ * CoreAudio audio output driver for macOS
  *
  * original copyright (C) Timothy J. Wood - Aug 2000
  * ported to MPlayer libao2 by Dan Christiansen
@@ -28,7 +28,7 @@
  */
 
 /*
- * The MacOS X CoreAudio framework doesn't mesh as simply as some
+ * The macOS CoreAudio framework doesn't mesh as simply as some
  * simpler frameworks do.  This is due to the fact that CoreAudio pulls
  * audio samples rather than having them pushed at it (which is nice
  * when you are wanting to do good buffering of audio).
@@ -114,7 +114,7 @@ static OSStatus enable_property_listener(struct ao *ao, bool enabled)
                             kAudioHardwarePropertyDevices};
     AudioDeviceID devs[] = {p->device,
                             kAudioObjectSystemObject};
-    assert(MP_ARRAY_SIZE(selectors) == MP_ARRAY_SIZE(devs));
+    static_assert(MP_ARRAY_SIZE(selectors) == MP_ARRAY_SIZE(devs), "");
 
     OSStatus status = noErr;
     for (int n = 0; n < MP_ARRAY_SIZE(devs); n++) {
diff --git a/audio/out/ao_coreaudio_properties.h b/audio/out/ao_coreaudio_properties.h
index f293968..2c9c565 100644
--- a/audio/out/ao_coreaudio_properties.h
+++ b/audio/out/ao_coreaudio_properties.h
@@ -23,6 +23,11 @@
 
 #include "internal.h"
 
+#if HAVE_AVFOUNDATION || HAVE_AUDIOUNIT
+#undef HAVE_COREAUDIO
+#define HAVE_COREAUDIO 1
+#endif
+
 // CoreAudio names are way too verbose
 #define ca_sel    AudioObjectPropertySelector
 #define ca_scope  AudioObjectPropertyScope
diff --git a/audio/out/ao_coreaudio_utils.c b/audio/out/ao_coreaudio_utils.c
index 14db8e3..e74092a 100644
--- a/audio/out/ao_coreaudio_utils.c
+++ b/audio/out/ao_coreaudio_utils.c
@@ -138,7 +138,8 @@ bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
 {
     if (code == noErr) return true;
 
-    mp_msg(ao->log, level, "%s (%s/%d)\n", message, mp_tag_str(code), (int)code);
+    if (ao)
+        mp_msg(ao->log, level, "%s (%s/%d)\n", message, mp_tag_str(code), (int)code);
 
     return false;
 }
@@ -470,11 +471,9 @@ bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
 
     ca_print_asbd(ao, "setting stream physical format:", &change_format);
 
-    sem_t wakeup;
-    if (mp_sem_init(&wakeup, 0, 0)) {
-        MP_WARN(ao, "OOM\n");
-        return false;
-    }
+    mp_sem_t wakeup;
+    if (mp_sem_init(&wakeup, 0, 0))
+        MP_HANDLE_OOM(0);
 
     AudioStreamBasicDescription prev_format;
     err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
diff --git a/audio/out/ao_coreaudio_utils.h b/audio/out/ao_coreaudio_utils.h
index 0e2b8b1..699ffde 100644
--- a/audio/out/ao_coreaudio_utils.h
+++ b/audio/out/ao_coreaudio_utils.h
@@ -27,7 +27,12 @@
 #include "common/msg.h"
 #include "audio/out/ao.h"
 #include "internal.h"
-#include "osdep/apple_utils.h"
+#include "osdep/utils-mac.h"
+
+#if HAVE_AVFOUNDATION || HAVE_AUDIOUNIT
+#undef HAVE_COREAUDIO
+#define HAVE_COREAUDIO 1
+#endif
 
 bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
 
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 163fdca..4bae438 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -26,6 +26,7 @@
 #include <limits.h>
 
 #include <libavutil/common.h>
+#include <libavutil/samplefmt.h>
 
 #include "config.h"
 #include "options/options.h"
diff --git a/audio/out/ao_null.c b/audio/out/ao_null.c
index fcb61d2..0cda8d9 100644
--- a/audio/out/ao_null.c
+++ b/audio/out/ao_null.c
@@ -116,10 +116,11 @@ static void uninit(struct ao *ao)
 {
 }
 
-// stop playing and empty buffers (for seeking/pause)
+// stop playing and empty buffers (for seeking)
 static void reset(struct ao *ao)
 {
     struct priv *priv = ao->priv;
+    priv->paused = false;
     priv->buffered = 0;
     priv->playing = false;
 }
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index 5c0b8c9..afe5839 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -36,6 +36,7 @@
 #include <sys/types.h>
 
 #include "audio/format.h"
+#include "common/common.h"
 #include "common/msg.h"
 #include "options/options.h"
 #include "osdep/endian.h"
@@ -89,7 +90,7 @@ static const int format_table[][2] = {
 
 #define MP_WARN_IOCTL_ERR(__ao) \
     MP_WARN((__ao), "%s: ioctl() fail, err = %i: %s\n", \
-        __FUNCTION__, errno, strerror(errno))
+        __FUNCTION__, errno, mp_strerror(errno))
 
 
 static void uninit(struct ao *ao);
@@ -329,7 +330,7 @@ static bool audio_write(struct ao *ao, void **data, int samples)
         if (errno == EINTR)
 			continue;
         MP_WARN(ao, "audio_write: write() fail, err = %i: %s.\n",
-            errno, strerror(errno));
+            errno, mp_strerror(errno));
         return false;
     }
     if ((size_t)rc != size) {
diff --git a/audio/out/ao_pipewire.c b/audio/out/ao_pipewire.c
index 3fbcbf6..94d393a 100644
--- a/audio/out/ao_pipewire.c
+++ b/audio/out/ao_pipewire.c
@@ -27,6 +27,7 @@
 #include <spa/utils/result.h>
 #include <math.h>
 
+#include "common/common.h"
 #include "common/msg.h"
 #include "options/m_config.h"
 #include "options/m_option.h"
@@ -46,6 +47,15 @@ static inline int pw_stream_get_time_n(struct pw_stream *stream, struct pw_time
 #define spa_hook_remove(hook) if ((hook)->link.prev) spa_hook_remove(hook)
 #endif
 
+#if !PW_CHECK_VERSION(1, 0, 4)
+static uint64_t pw_stream_get_nsec(struct pw_stream *stream)
+{
+	struct timespec ts;
+	clock_gettime(CLOCK_MONOTONIC, &ts);
+	return SPA_TIMESPEC_TO_NSEC(&ts);
+}
+#endif
+
 enum init_state {
     INIT_STATE_NONE,
     INIT_STATE_SUCCESS,
@@ -86,7 +96,7 @@ struct id_list {
     struct spa_list node;
 };
 
-static enum spa_audio_format af_fmt_to_pw(struct ao *ao, enum af_format format)
+static enum spa_audio_format af_fmt_to_pw(enum af_format format)
 {
     switch (format) {
     case AF_FORMAT_U8:          return SPA_AUDIO_FORMAT_U8;
@@ -99,9 +109,21 @@ static enum spa_audio_format af_fmt_to_pw(struct ao *ao, enum af_format format)
     case AF_FORMAT_S32P:        return SPA_AUDIO_FORMAT_S32P;
     case AF_FORMAT_FLOATP:      return SPA_AUDIO_FORMAT_F32P;
     case AF_FORMAT_DOUBLEP:     return SPA_AUDIO_FORMAT_F64P;
-    default:
-                                MP_WARN(ao, "Unhandled format %d\n", format);
-                                return SPA_AUDIO_FORMAT_UNKNOWN;
+    default:                    return SPA_AUDIO_FORMAT_UNKNOWN;
+    }
+}
+
+static enum spa_audio_iec958_codec af_fmt_to_codec(enum af_format format)
+{
+    switch (format) {
+    case AF_FORMAT_S_AAC:    return SPA_AUDIO_IEC958_CODEC_MPEG2_AAC;
+    case AF_FORMAT_S_AC3:    return SPA_AUDIO_IEC958_CODEC_AC3;
+    case AF_FORMAT_S_DTS:    return SPA_AUDIO_IEC958_CODEC_DTS;
+    case AF_FORMAT_S_DTSHD:  return SPA_AUDIO_IEC958_CODEC_DTSHD;
+    case AF_FORMAT_S_EAC3:   return SPA_AUDIO_IEC958_CODEC_EAC3;
+    case AF_FORMAT_S_MP3:    return SPA_AUDIO_IEC958_CODEC_MPEG;
+    case AF_FORMAT_S_TRUEHD: return SPA_AUDIO_IEC958_CODEC_TRUEHD;
+    default:                 return SPA_AUDIO_IEC958_CODEC_UNKNOWN;
     }
 }
 
@@ -154,14 +176,13 @@ static void on_process(void *userdata)
     void *data[MP_NUM_CHANNELS];
 
     if ((b = pw_stream_dequeue_buffer(p->stream)) == NULL) {
-        MP_WARN(ao, "out of buffers: %s\n", strerror(errno));
+        MP_WARN(ao, "out of buffers: %s\n", mp_strerror(errno));
         return;
     }
 
     struct spa_buffer *buf = b->buffer;
 
-    int bytes_per_channel = buf->datas[0].maxsize / ao->channels.num;
-    int nframes = bytes_per_channel / ao->sstride;
+    int nframes = buf->datas[0].maxsize / ao->sstride;
 #if PW_CHECK_VERSION(0, 3, 49)
     if (b->requested != 0)
         nframes = MPMIN(b->requested, nframes);
@@ -177,9 +198,13 @@ static void on_process(void *userdata)
         time.rate.num = 1;
 
     int64_t end_time = mp_time_ns();
-    /* time.queued is always going to be 0, so we don't need to care */
-    end_time += (nframes * 1e9 / ao->samplerate) +
-                ((double) time.delay * SPA_NSEC_PER_SEC * time.rate.num / time.rate.denom);
+    end_time += MP_TIME_S_TO_NS(nframes) / ao->samplerate;
+    end_time += MP_TIME_S_TO_NS(time.delay) * time.rate.num / time.rate.denom;
+    end_time += MP_TIME_S_TO_NS(time.queued) / ao->samplerate;
+#if PW_CHECK_VERSION(0, 3, 50)
+    end_time += MP_TIME_S_TO_NS(time.buffered) / ao->samplerate;
+#endif
+    end_time -= pw_stream_get_nsec(p->stream) - time.now;
 
     int samples = ao_read_data_nonblocking(ao, data, nframes, end_time);
     b->size = samples;
@@ -214,7 +239,7 @@ static void on_param_changed(void *userdata, uint32_t id, const struct spa_pod *
     if (param == NULL || id != SPA_PARAM_Format)
         return;
 
-    int buffer_size = ao->device_buffer * af_fmt_to_bytes(ao->format) * ao->channels.num;
+    int buffer_size = ao->device_buffer * ao->sstride;
 
     params[0] = spa_pod_builder_add_object(&b,
                     SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
@@ -506,7 +531,7 @@ static int pipewire_init_boilerplate(struct ao *ao)
     if (!p->core) {
         MP_MSG(ao, ao->probing ? MSGL_V : MSGL_ERR,
                "Could not connect to context '%s': %s\n",
-               p->options.remote, strerror(errno));
+               p->options.remote, mp_strerror(errno));
         pw_context_destroy(context);
         goto error;
     }
@@ -580,24 +605,41 @@ static int init(struct ao *ao)
 
     pw_properties_setf(props, PW_KEY_NODE_RATE, "1/%d", ao->samplerate);
 
-    enum spa_audio_format spa_format = af_fmt_to_pw(ao, ao->format);
-    if (spa_format == SPA_AUDIO_FORMAT_UNKNOWN) {
-        ao->format = AF_FORMAT_FLOATP;
-        spa_format = SPA_AUDIO_FORMAT_F32P;
-    }
+    if (af_fmt_is_spdif(ao->format)) {
+        enum spa_audio_iec958_codec spa_codec = af_fmt_to_codec(ao->format);
+        if (spa_codec == SPA_AUDIO_IEC958_CODEC_UNKNOWN) {
+            MP_ERR(ao, "Unhandled codec %d\n", ao->format);
+            goto error_props;
+        }
 
-    struct spa_audio_info_raw audio_info = {
-        .format = spa_format,
-        .rate = ao->samplerate,
-        .channels = ao->channels.num,
-    };
+        struct spa_audio_info_iec958 audio_info = {
+            .codec = spa_codec,
+            .rate = ao->samplerate,
+        };
 
-    for (int i = 0; i < ao->channels.num; i++)
-        audio_info.position[i] = mp_speaker_id_to_spa(ao, ao->channels.speaker[i]);
+        params[0] = spa_format_audio_iec958_build(&b, SPA_PARAM_EnumFormat, &audio_info);
+        if (!params[0])
+            goto error_props;
+    } else {
+        enum spa_audio_format spa_format = af_fmt_to_pw(ao->format);
+        if (spa_format == SPA_AUDIO_FORMAT_UNKNOWN) {
+            MP_ERR(ao, "Unhandled format %d\n", ao->format);
+            goto error_props;
+        }
 
-    params[0] = spa_format_audio_raw_build(&b, SPA_PARAM_EnumFormat, &audio_info);
-    if (!params[0])
-        goto error_props;
+        struct spa_audio_info_raw audio_info = {
+            .format = spa_format,
+            .rate = ao->samplerate,
+            .channels = ao->channels.num,
+        };
+
+        for (int i = 0; i < ao->channels.num; i++)
+            audio_info.position[i] = mp_speaker_id_to_spa(ao, ao->channels.speaker[i]);
+
+        params[0] = spa_format_audio_raw_build(&b, SPA_PARAM_EnumFormat, &audio_info);
+        if (!params[0])
+            goto error_props;
+    }
 
     if (af_fmt_is_planar(ao->format)) {
         ao->num_planes = ao->channels.num;
@@ -664,6 +706,15 @@ static void start(struct ao *ao)
     pw_thread_loop_unlock(p->loop);
 }
 
+static bool set_pause(struct ao *ao, bool paused)
+{
+    struct priv *p = ao->priv;
+    pw_thread_loop_lock(p->loop);
+    pw_stream_set_active(p->stream, !paused);
+    pw_thread_loop_unlock(p->loop);
+    return true;
+}
+
 #define CONTROL_RET(r) (!r ? CONTROL_OK : CONTROL_ERROR)
 
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
@@ -855,7 +906,7 @@ const struct ao_driver audio_out_pipewire = {
     .uninit      = uninit,
     .reset       = reset,
     .start       = start,
-
+    .set_pause   = set_pause,
     .control     = control,
 
     .hotplug_init   = hotplug_init,
diff --git a/audio/out/ao_pulse.c b/audio/out/ao_pulse.c
index 3b29b1a..5c86855 100644
--- a/audio/out/ao_pulse.c
+++ b/audio/out/ao_pulse.c
@@ -118,7 +118,7 @@ static void stream_request_cb(pa_stream *s, size_t length, void *userdata)
 {
     struct ao *ao = userdata;
     struct priv *priv = ao->priv;
-    ao_wakeup_playthread(ao);
+    ao_wakeup(ao);
     pa_threaded_mainloop_signal(priv->mainloop, 0);
 }
 
@@ -135,7 +135,7 @@ static void underflow_cb(pa_stream *s, void *userdata)
     struct priv *priv = ao->priv;
     priv->playing = false;
     priv->underrun_signalled = true;
-    ao_wakeup_playthread(ao);
+    ao_wakeup(ao);
     pa_threaded_mainloop_signal(priv->mainloop, 0);
 }
 
@@ -804,6 +804,7 @@ const struct ao_driver audio_out_pulse = {
     .priv_size = sizeof(struct priv),
     .priv_defaults = &(const struct priv) {
         .cfg_buffer = 100,
+        .cfg_latency_hacks = true,
     },
     .options = (const struct m_option[]) {
         {"host", OPT_STRING(cfg_host)},
diff --git a/audio/out/ao_sndio.c b/audio/out/ao_sndio.c
index fce7139..309ea4a 100644
--- a/audio/out/ao_sndio.c
+++ b/audio/out/ao_sndio.c
@@ -22,6 +22,8 @@
 #include <errno.h>
 #include <sndio.h>
 
+#include "config.h"
+
 #include "options/m_option.h"
 #include "common/msg.h"
 
@@ -292,7 +294,7 @@ static void get_state(struct ao *ao, struct mp_pcm_state *state)
     state->delay = p->delay / (double)p->par.rate;
 
     /* report unexpected EOF / underrun */
-    if ((state->queued_samples && state->queued_samples &&
+    if ((state->queued_samples &&
         (state->queued_samples < state->free_samples) &&
         p->playing) || sio_eof(p->hdl))
     {
@@ -301,7 +303,7 @@ static void get_state(struct ao *ao, struct mp_pcm_state *state)
                 state->free_samples, state->queued_samples, state->delay);
         p->playing = false;
         state->playing = p->playing;
-        ao_wakeup_playthread(ao);
+        ao_wakeup(ao);
     } else {
         state->playing = p->playing;
     }
diff --git a/audio/out/ao_wasapi.c b/audio/out/ao_wasapi.c
index b201f26..d986d80 100644
--- a/audio/out/ao_wasapi.c
+++ b/audio/out/ao_wasapi.c
@@ -150,14 +150,32 @@ exit_label:
     return false;
 }
 
+static void thread_pause(struct ao *ao)
+{
+    struct wasapi_state *state = ao->priv;
+    MP_DBG(state, "Thread Pause\n");
+    HRESULT hr = IAudioClient_Stop(state->pAudioClient);
+    if (FAILED(hr))
+        MP_ERR(state, "IAudioClient_Stop returned: %s\n", mp_HRESULT_to_str(hr));
+}
+
+static void thread_unpause(struct ao *ao)
+{
+    struct wasapi_state *state = ao->priv;
+    MP_DBG(state, "Thread Unpause\n");
+    HRESULT hr = IAudioClient_Start(state->pAudioClient);
+    if (FAILED(hr)) {
+        MP_ERR(state, "IAudioClient_Start returned %s\n",
+               mp_HRESULT_to_str(hr));
+    }
+}
+
 static void thread_reset(struct ao *ao)
 {
     struct wasapi_state *state = ao->priv;
     HRESULT hr;
     MP_DBG(state, "Thread Reset\n");
-    hr = IAudioClient_Stop(state->pAudioClient);
-    if (FAILED(hr))
-        MP_ERR(state, "IAudioClient_Stop returned: %s\n", mp_HRESULT_to_str(hr));
+    thread_pause(ao);
 
     hr = IAudioClient_Reset(state->pAudioClient);
     if (FAILED(hr))
@@ -172,27 +190,20 @@ static void thread_resume(struct ao *ao)
     MP_DBG(state, "Thread Resume\n");
     thread_reset(ao);
     thread_feed(ao);
-
-    HRESULT hr = IAudioClient_Start(state->pAudioClient);
-    if (FAILED(hr)) {
-        MP_ERR(state, "IAudioClient_Start returned %s\n",
-               mp_HRESULT_to_str(hr));
-    }
+    thread_unpause(ao);
 }
 
-static void thread_wakeup(void *ptr)
+static void set_state_and_wakeup_thread(struct ao *ao,
+                                        enum wasapi_thread_state thread_state)
 {
-    struct ao *ao = ptr;
     struct wasapi_state *state = ao->priv;
+    atomic_store(&state->thread_state, thread_state);
     SetEvent(state->hWake);
 }
 
-static void set_thread_state(struct ao *ao,
-                             enum wasapi_thread_state thread_state)
+static void thread_process_dispatch(void *ptr)
 {
-    struct wasapi_state *state = ao->priv;
-    atomic_store(&state->thread_state, thread_state);
-    thread_wakeup(ao);
+    set_state_and_wakeup_thread(ptr, WASAPI_THREAD_DISPATCH);
 }
 
 static DWORD __stdcall AudioThread(void *lpParameter)
@@ -212,8 +223,6 @@ static DWORD __stdcall AudioThread(void *lpParameter)
         if (WaitForSingleObject(state->hWake, INFINITE) != WAIT_OBJECT_0)
             MP_ERR(ao, "Unexpected return value from WaitForSingleObject\n");
 
-        mp_dispatch_queue_process(state->dispatch, 0);
-
         int thread_state = atomic_load(&state->thread_state);
         switch (thread_state) {
         case WASAPI_THREAD_FEED:
@@ -221,6 +230,9 @@ static DWORD __stdcall AudioThread(void *lpParameter)
             if (thread_feed(ao) && thread_feed(ao))
                 MP_ERR(ao, "Unable to fill buffer fast enough\n");
             break;
+        case WASAPI_THREAD_DISPATCH:
+            mp_dispatch_queue_process(state->dispatch, 0);
+            break;
         case WASAPI_THREAD_RESET:
             thread_reset(ao);
             break;
@@ -230,6 +242,12 @@ static DWORD __stdcall AudioThread(void *lpParameter)
         case WASAPI_THREAD_SHUTDOWN:
             thread_reset(ao);
             goto exit_label;
+        case WASAPI_THREAD_PAUSE:
+            thread_pause(ao);
+            break;
+        case WASAPI_THREAD_UNPAUSE:
+            thread_unpause(ao);
+            break;
         default:
             MP_ERR(ao, "Unhandled thread state: %d\n", thread_state);
         }
@@ -250,7 +268,7 @@ static void uninit(struct ao *ao)
     MP_DBG(ao, "Uninit wasapi\n");
     struct wasapi_state *state = ao->priv;
     if (state->hWake)
-        set_thread_state(ao, WASAPI_THREAD_SHUTDOWN);
+        set_state_and_wakeup_thread(ao, WASAPI_THREAD_SHUTDOWN);
 
     if (state->hAudioThread &&
         WaitForSingleObject(state->hAudioThread, INFINITE) != WAIT_OBJECT_0)
@@ -301,7 +319,7 @@ static int init(struct ao *ao)
     }
 
     state->dispatch = mp_dispatch_create(state);
-    mp_dispatch_set_wakeup_fn(state->dispatch, thread_wakeup, ao);
+    mp_dispatch_set_wakeup_fn(state->dispatch, thread_process_dispatch, ao);
 
     state->init_ok = false;
     state->hAudioThread = CreateThread(NULL, 0, &AudioThread, ao, 0, NULL);
@@ -349,7 +367,7 @@ static int thread_control_exclusive(struct ao *ao, enum aocontrol cmd, void *arg
     case AOCONTROL_GET_VOLUME:
         IAudioEndpointVolume_GetMasterVolumeLevelScalar(
             state->pEndpointVolume, &volume);
-        *(float *)arg = volume;
+        *(float *)arg = volume * 100.f;
         return CONTROL_OK;
     case AOCONTROL_SET_VOLUME:
         volume = (*(float *)arg) / 100.f;
@@ -379,7 +397,7 @@ static int thread_control_shared(struct ao *ao, enum aocontrol cmd, void *arg)
     switch(cmd) {
     case AOCONTROL_GET_VOLUME:
         ISimpleAudioVolume_GetMasterVolume(state->pAudioVolume, &volume);
-        *(float *)arg = volume;
+        *(float *)arg = volume * 100.f;
         return CONTROL_OK;
     case AOCONTROL_SET_VOLUME:
         volume = (*(float *)arg) / 100.f;
@@ -456,12 +474,18 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 
 static void audio_reset(struct ao *ao)
 {
-    set_thread_state(ao, WASAPI_THREAD_RESET);
+    set_state_and_wakeup_thread(ao, WASAPI_THREAD_RESET);
 }
 
 static void audio_resume(struct ao *ao)
 {
-    set_thread_state(ao, WASAPI_THREAD_RESUME);
+    set_state_and_wakeup_thread(ao, WASAPI_THREAD_RESUME);
+}
+
+static bool audio_set_pause(struct ao *ao, bool paused)
+{
+    set_state_and_wakeup_thread(ao, paused ? WASAPI_THREAD_PAUSE : WASAPI_THREAD_UNPAUSE);
+    return true;
 }
 
 static void hotplug_uninit(struct ao *ao)
@@ -497,6 +521,7 @@ const struct ao_driver audio_out_wasapi = {
     .control        = control,
     .reset          = audio_reset,
     .start          = audio_resume,
+    .set_pause      = audio_set_pause,
     .list_devs      = wasapi_list_devs,
     .hotplug_init   = hotplug_init,
     .hotplug_uninit = hotplug_uninit,
diff --git a/audio/out/ao_wasapi.h b/audio/out/ao_wasapi.h
index 17b8f7a..4e5e9c8 100644
--- a/audio/out/ao_wasapi.h
+++ b/audio/out/ao_wasapi.h
@@ -48,9 +48,12 @@ void wasapi_change_uninit(struct ao* ao);
 
 enum wasapi_thread_state {
     WASAPI_THREAD_FEED = 0,
+    WASAPI_THREAD_DISPATCH,
     WASAPI_THREAD_RESUME,
     WASAPI_THREAD_RESET,
-    WASAPI_THREAD_SHUTDOWN
+    WASAPI_THREAD_SHUTDOWN,
+    WASAPI_THREAD_PAUSE,
+    WASAPI_THREAD_UNPAUSE,
 };
 
 typedef struct wasapi_state {
diff --git a/audio/out/ao_wasapi_utils.c b/audio/out/ao_wasapi_utils.c
index 731fe8a..7e85f75 100644
--- a/audio/out/ao_wasapi_utils.c
+++ b/audio/out/ao_wasapi_utils.c
@@ -18,12 +18,15 @@
  */
 
 #include <math.h>
-#include <wchar.h>
+
 #include <windows.h>
-#include <errors.h>
+#include <mmsystem.h>
+#include <mmreg.h>
 #include <ksguid.h>
 #include <ksmedia.h>
 #include <avrt.h>
+#include <propsys.h>
+#include <functiondiscoverykeys_devpkey.h>
 
 #include "audio/format.h"
 #include "osdep/timer.h"
@@ -31,36 +34,47 @@
 #include "osdep/strnlen.h"
 #include "ao_wasapi.h"
 
-DEFINE_PROPERTYKEY(mp_PKEY_Device_FriendlyName,
-                   0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20,
-                   0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 14);
-DEFINE_PROPERTYKEY(mp_PKEY_Device_DeviceDesc,
-                   0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20,
-                   0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 2);
-// CEA 861 subformats
-// should work on vista
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_DTS,
-            0x00000008, 0x0000, 0x0010, 0x80, 0x00,
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_DTS
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_DTS,
+            WAVE_FORMAT_DTS, 0x0000, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL,
-            0x00000092, 0x0000, 0x0010, 0x80, 0x00,
+#endif
+
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL,
+            WAVE_FORMAT_DOLBY_AC3_SPDIF, 0x0000, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-// might require 7+
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_AAC,
+#endif
+
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_AAC
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_AAC,
             0x00000006, 0x0cea, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_MPEG3,
-            0x00000004, 0x0cea, 0x0010, 0x80, 0x00,
+#endif
+
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_MPEG3
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_MPEG3,
+            0x00000005, 0x0cea, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS,
+#endif
+
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS,
             0x0000000a, 0x0cea, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD,
+#endif
+
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD,
             0x0000000b, 0x0cea, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
-DEFINE_GUID(mp_KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP,
+#endif
+
+#ifndef KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP,
             0x0000000c, 0x0cea, 0x0010, 0x80, 0x00,
             0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
+#endif
 
 struct wasapi_sample_fmt {
     int mp_format;  // AF_FORMAT_*
@@ -82,13 +96,13 @@ static const struct wasapi_sample_fmt wasapi_formats[] = {
     // aka S24 (with conversion on output)
     {AF_FORMAT_S32,     24, 24, &KSDATAFORMAT_SUBTYPE_PCM},
     {AF_FORMAT_FLOAT,   32, 32, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT},
-    {AF_FORMAT_S_AC3,   16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL},
-    {AF_FORMAT_S_DTS,   16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_DTS},
-    {AF_FORMAT_S_AAC,   16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_AAC},
-    {AF_FORMAT_S_MP3,   16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_MPEG3},
-    {AF_FORMAT_S_TRUEHD, 16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP},
-    {AF_FORMAT_S_EAC3,  16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS},
-    {AF_FORMAT_S_DTSHD, 16, 16, &mp_KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD},
+    {AF_FORMAT_S_AC3,   16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL},
+    {AF_FORMAT_S_DTS,   16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_DTS},
+    {AF_FORMAT_S_AAC,   16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_AAC},
+    {AF_FORMAT_S_MP3,   16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_MPEG3},
+    {AF_FORMAT_S_TRUEHD, 16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP},
+    {AF_FORMAT_S_EAC3,  16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS},
+    {AF_FORMAT_S_DTSHD, 16, 16, &KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD},
     {0},
 };
 
@@ -562,9 +576,10 @@ static void init_session_display(struct wasapi_state *state, const char *name) {
                                          (void **)&state->pSessionControl);
     EXIT_ON_ERROR(hr);
 
-    wchar_t path[MAX_PATH] = {0};
-    GetModuleFileNameW(NULL, path, MAX_PATH);
+    wchar_t *path = talloc_array(NULL, wchar_t, MP_PATH_MAX);
+    GetModuleFileNameW(NULL, path, MP_PATH_MAX);
     hr = IAudioSessionControl_SetIconPath(state->pSessionControl, path, NULL);
+    talloc_free(path);
     if (FAILED(hr)) {
         // don't goto exit_label here since SetDisplayName might still work
         MP_WARN(state, "Error setting audio session icon: %s\n",
@@ -718,7 +733,7 @@ static char* get_device_name(struct mp_log *l, void *talloc_ctx, IMMDevice *pDev
     HRESULT hr = IMMDevice_OpenPropertyStore(pDevice, STGM_READ, &pProps);
     EXIT_ON_ERROR(hr);
 
-    hr = IPropertyStore_GetValue(pProps, &mp_PKEY_Device_FriendlyName,
+    hr = IPropertyStore_GetValue(pProps, &PKEY_Device_FriendlyName,
                                  &devname);
     EXIT_ON_ERROR(hr);
 
diff --git a/audio/out/buffer.c b/audio/out/buffer.c
index 5b8b523..97f7ea1 100644
--- a/audio/out/buffer.c
+++ b/audio/out/buffer.c
@@ -41,7 +41,7 @@ struct buffer_state {
     mp_mutex lock;
     mp_cond wakeup;
 
-    // Playthread sleep
+    // AO thread sleep
     mp_mutex pt_lock;
     mp_cond pt_wakeup;
 
@@ -62,6 +62,11 @@ struct buffer_state {
     bool paused;                // logically paused
 
     int64_t end_time_ns;        // absolute output time of last played sample
+    int64_t queued_time_ns;     // duration of samples that have been queued to
+                                // the device but have not been played.
+                                // This field is only set in ao_set_paused(),
+                                // and is considered as a temporary solution;
+                                // DO NOT USE IT IN OTHER PLACES.
 
     bool initial_unblocked;
 
@@ -78,9 +83,9 @@ struct buffer_state {
     bool terminate;             // exit thread
 };
 
-static MP_THREAD_VOID playthread(void *arg);
+static MP_THREAD_VOID ao_thread(void *arg);
 
-void ao_wakeup_playthread(struct ao *ao)
+void ao_wakeup(struct ao *ao)
 {
     struct buffer_state *p = ao->buffer_state;
     mp_mutex_lock(&p->pt_lock);
@@ -173,8 +178,8 @@ static int read_buffer(struct ao *ao, void **data, int samples, bool *eof,
     return pos;
 }
 
-static int ao_read_data_unlocked(struct ao *ao, void **data, int samples,
-                                 int64_t out_time_ns, bool pad_silence)
+static int ao_read_data_locked(struct ao *ao, void **data, int samples,
+                               int64_t out_time_ns, bool pad_silence)
 {
     struct buffer_state *p = ao->buffer_state;
     assert(!ao->driver->write);
@@ -208,7 +213,7 @@ int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_ns)
 
     mp_mutex_lock(&p->lock);
 
-    int pos = ao_read_data_unlocked(ao, data, samples, out_time_ns, true);
+    int pos = ao_read_data_locked(ao, data, samples, out_time_ns, true);
 
     mp_mutex_unlock(&p->lock);
 
@@ -224,7 +229,7 @@ int ao_read_data_nonblocking(struct ao *ao, void **data, int samples, int64_t ou
     if (mp_mutex_trylock(&p->lock))
             return 0;
 
-    int pos = ao_read_data_unlocked(ao, data, samples, out_time_ns, false);
+    int pos = ao_read_data_locked(ao, data, samples, out_time_ns, false);
 
     mp_mutex_unlock(&p->lock);
 
@@ -347,7 +352,7 @@ void ao_reset(struct ao *ao)
         ao->driver->reset(ao);
 
     if (wakeup)
-        ao_wakeup_playthread(ao);
+        ao_wakeup(ao);
 }
 
 // Initiate playback. This moves from the stop/underrun state to actually
@@ -374,14 +379,14 @@ void ao_start(struct ao *ao)
     if (do_start)
         ao->driver->start(ao);
 
-    ao_wakeup_playthread(ao);
+    ao_wakeup(ao);
 }
 
 void ao_set_paused(struct ao *ao, bool paused, bool eof)
 {
     struct buffer_state *p = ao->buffer_state;
     bool wakeup = false;
-    bool do_reset = false, do_start = false;
+    bool do_change_state = false;
 
     // If we are going to pause on eof and ao is still playing,
     // be sure to drain the ao first for gapless.
@@ -402,9 +407,9 @@ void ao_set_paused(struct ao *ao, bool paused, bool eof)
                     p->streaming = false;
                     p->recover_pause = !ao->untimed;
                 }
-            } else if (ao->driver->reset) {
+            } else if (ao->driver->reset || ao->driver->set_pause) {
                 // See ao_reset() why this is done outside of the lock.
-                do_reset = true;
+                do_change_state = true;
                 p->streaming = false;
             }
         }
@@ -416,7 +421,7 @@ void ao_set_paused(struct ao *ao, bool paused, bool eof)
             p->hw_paused = false;
         } else {
             if (!p->streaming)
-                do_start = true;
+                do_change_state = true;
             p->streaming = true;
         }
         wakeup = true;
@@ -425,13 +430,25 @@ void ao_set_paused(struct ao *ao, bool paused, bool eof)
 
     mp_mutex_unlock(&p->lock);
 
-    if (do_reset)
-        ao->driver->reset(ao);
-    if (do_start)
-        ao->driver->start(ao);
+    if (do_change_state) {
+        if (ao->driver->set_pause) {
+            if (paused) {
+                ao->driver->set_pause(ao, true);
+                p->queued_time_ns = p->end_time_ns - mp_time_ns();
+            } else {
+                p->end_time_ns = p->queued_time_ns + mp_time_ns();
+                ao->driver->set_pause(ao, false);
+            }
+        } else {
+            if (paused)
+                ao->driver->reset(ao);
+            else
+                ao->driver->start(ao);
+        }
+    }
 
     if (wakeup)
-        ao_wakeup_playthread(ao);
+        ao_wakeup(ao);
 }
 
 // Whether audio is playing. This means that there is still data in the buffers,
@@ -486,7 +503,7 @@ void ao_drain(struct ao *ao)
 static void wakeup_filters(void *ctx)
 {
     struct ao *ao = ctx;
-    ao_wakeup_playthread(ao);
+    ao_wakeup(ao);
 }
 
 void ao_uninit(struct ao *ao)
@@ -561,7 +578,7 @@ bool init_buffer_post(struct ao *ao)
         mp_filter_graph_set_wakeup_cb(p->filter_root, wakeup_filters, ao);
 
         p->thread_valid = true;
-        if (mp_thread_create(&p->thread, playthread, ao)) {
+        if (mp_thread_create(&p->thread, ao_thread, ao)) {
             p->thread_valid = false;
             return false;
         }
@@ -684,7 +701,7 @@ eof:
     return true;
 }
 
-static MP_THREAD_VOID playthread(void *arg)
+static MP_THREAD_VOID ao_thread(void *arg)
 {
     struct ao *ao = arg;
     struct buffer_state *p = ao->buffer_state;
@@ -731,6 +748,6 @@ void ao_unblock(struct ao *ao)
         mp_mutex_lock(&p->lock);
         p->initial_unblocked = true;
         mp_mutex_unlock(&p->lock);
-        ao_wakeup_playthread(ao);
+        ao_wakeup(ao);
     }
 }
diff --git a/audio/out/internal.h b/audio/out/internal.h
index 7951b38..51429b9 100644
--- a/audio/out/internal.h
+++ b/audio/out/internal.h
@@ -108,6 +108,7 @@ struct mp_pcm_state {
  *          start
  *     Optional for both types:
  *          control
+ *          set_pause
  *  a) ->write is called to queue audio. push.c creates a thread to regularly
  *     refill audio device buffers with ->write, but all driver functions are
  *     always called under an exclusive lock.
@@ -115,8 +116,6 @@ struct mp_pcm_state {
  *          reset
  *          write
  *          get_state
- *     Optional:
- *          set_pause
  *  b) ->write must be NULL. ->start must be provided, and should make the
  *     audio API start calling the audio callback. Your audio callback should
  *     in turn call ao_read_data() to get audio data. Most functions are
@@ -149,6 +148,9 @@ struct ao_driver {
     // Stop all audio playback, clear buffers, back to state after init().
     // Optional for pull AOs.
     void (*reset)(struct ao *ao);
+    // pull based: set pause state. Only called after start() and before reset().
+    //             The return value is ignored.
+    //             The pausing state is also cleared by reset().
     // push based: set pause state. Only called after start() and before reset().
     //             returns success (this is intended for paused=true; if it
     //             returns false, playback continues, and the core emulates via
@@ -157,7 +159,7 @@ struct ao_driver {
     bool (*set_pause)(struct ao *ao, bool paused);
     // pull based: start the audio callback
     // push based: start playing queued data
-    //             AO should call ao_wakeup_playthread() if a period boundary
+    //             AO should call ao_wakeup() if a period boundary
     //             is crossed, or playback stops due to external reasons
     //             (including underruns or device removal)
     //             must set mp_pcm_state.playing; unset on error/underrun/end
@@ -229,7 +231,7 @@ bool ao_can_convert_inplace(struct ao_convert_fmt *fmt);
 bool ao_need_conversion(struct ao_convert_fmt *fmt);
 void ao_convert_inplace(struct ao_convert_fmt *fmt, void **data, int num_samples);
 
-void ao_wakeup_playthread(struct ao *ao);
+void ao_wakeup(struct ao *ao);
 
 int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
                            void **data, int samples, int64_t out_time_ns);