ao_coreaudio: move spdif code to a new AO

The mplayer1/2/mpv CoreAudio audio output historically contained both usage
of AUHAL APIs (these go through the CoreAudio audio server) and the Device
based APIs (used only for output of compressed formats in exclusive mode).

The latter is a very unwieldy and low level API and pretty much forces us to
write a lot of code for little workr. Also with the widespread of HDMI, the
actual need for outputting compressed audio directly to the device is getting
lower (it was very useful with S/PDIF for bandwidth constraints not allowing
a number if channels transmitted in LPCM).

Considering how invasive it is (uses hog/exclusive mode), the new AO
(`ao_coreaudio_device`) is not going to be autoprobed but the user will have
to select it.

5 files changed, 595 insertions, 433 deletions

diff --git a/audio/out/ao.c b/audio/out/ao.c
index 378197bcb3..977b8eb69a 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -38,6 +38,7 @@
 
 extern const struct ao_driver audio_out_oss;
 extern const struct ao_driver audio_out_coreaudio;
+extern const struct ao_driver audio_out_coreaudio_exclusive;
 extern const struct ao_driver audio_out_rsound;
 extern const struct ao_driver audio_out_sndio;
 extern const struct ao_driver audio_out_pulse;
@@ -94,6 +95,9 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_ENCODING
     &audio_out_lavc,
 #endif
+#if HAVE_COREAUDIO
+    &audio_out_coreaudio_exclusive,
+#endif
 #if HAVE_RSOUND
     &audio_out_rsound,
 #endif
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index 1d44c81124..63009f3314 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -1,38 +1,18 @@
 /*
- * CoreAudio audio output driver for Mac OS X
+ * This file is part of mpv.
  *
- * original copyright (C) Timothy J. Wood - Aug 2000
- * ported to MPlayer libao2 by Dan Christiansen
- *
- * Chris Roccati
- * Stefano Pigozzi
- *
- * The S/PDIF part of the code is based on the auhal audio output
- * module from VideoLAN:
- * Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
+ * mpv is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
  * the Free Software Foundation; either version 2 of the License, or
  * (at your option) any later version.
  *
- * MPlayer is distributed in the hope that it will be useful,
+ * 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 General Public License for more details.
  *
  * You should have received a copy of the GNU General Public License along
- * along with MPlayer; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * The MacOS X 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).
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
  */
 
 #include "config.h"
@@ -50,47 +30,11 @@ static void audio_pause(struct ao *ao);
 static void audio_resume(struct ao *ao);
 static void reset(struct ao *ao);
 
-static void print_buffer(struct ao *ao, struct mp_ring *buffer)
-{
-    void *tctx = talloc_new(NULL);
-    MP_VERBOSE(ao, "%s\n", mp_ring_repr(buffer, tctx));
-    talloc_free(tctx);
-}
-
-struct priv_d {
-    // digital render callback
-    AudioDeviceIOProcID render_cb;
-
-    // pid set for hog mode, (-1) means that hog mode on the device was
-    // released. hog mode is exclusive access to a device
-    pid_t hog_pid;
-
-    // stream selected for digital playback by the detection in init
-    AudioStreamID stream;
-
-    // stream index in an AudioBufferList
-    int stream_idx;
-
-    // format we changed the stream to: for the digital case each application
-    // sets the stream format for a device to what it needs
-    AudioStreamBasicDescription stream_asbd;
-    AudioStreamBasicDescription original_asbd;
-
-    bool changed_mixing;
-    int stream_asbd_changed;
-    bool muted;
-};
-
 struct priv {
     AudioDeviceID device;   // selected device
-    bool is_digital;        // running in digital mode?
-
     AudioUnit audio_unit;   // AudioUnit for lpcm output
-
     bool paused;
-
     struct mp_ring *buffer;
-    struct priv_d *digital;
 
     // options
     int opt_device_id;
@@ -124,77 +68,29 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
     return noErr;
 }
 
-static OSStatus render_cb_digital(
-        AudioDeviceID device, const AudioTimeStamp *ts,
-        const void *in_data, const AudioTimeStamp *in_ts,
-        AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
-{
-    struct ao *ao    = ctx;
-    struct priv *p   = ao->priv;
-    struct priv_d *d = p->digital;
-    AudioBuffer buf  = out_data->mBuffers[d->stream_idx];
-    int requested    = buf.mDataByteSize;
-
-    if (d->muted)
-        mp_ring_drain(p->buffer, requested);
-    else
-        mp_ring_read(p->buffer, buf.mData, requested);
-
-    return noErr;
-}
-
 static int control(struct ao *ao, enum aocontrol cmd, void *arg)
 {
     struct priv *p = ao->priv;
-    ao_control_vol_t *control_vol;
-    OSStatus err;
-    Float32 vol;
     switch (cmd) {
-    case AOCONTROL_GET_VOLUME:
-        control_vol = (ao_control_vol_t *)arg;
-        if (p->is_digital) {
-            struct priv_d *d = p->digital;
-            // Digital output has no volume adjust.
-            int digitalvol = d->muted ? 0 : 100;
-            *control_vol = (ao_control_vol_t) {
-                .left = digitalvol, .right = digitalvol,
-            };
-            return CONTROL_TRUE;
-        }
-
-        err = AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume,
-                                    kAudioUnitScope_Global, 0, &vol);
-
+    case AOCONTROL_GET_VOLUME: {
+        ao_control_vol_t *control_vol = (ao_control_vol_t *)arg;
+        float vol;
+        OSStatus err =
+            AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume,
+                                  kAudioUnitScope_Global, 0, &vol);
         CHECK_CA_ERROR("could not get HAL output volume");
         control_vol->left = control_vol->right = vol * 100.0;
         return CONTROL_TRUE;
-
-    case AOCONTROL_SET_VOLUME:
-        control_vol = (ao_control_vol_t *)arg;
-
-        if (p->is_digital) {
-            struct priv_d *d = p->digital;
-            // Digital output can not set volume. Here we have to return true
-            // to make mixer forget it. Else mixer will add a soft filter,
-            // that's not we expected and the filter not support ac3 stream
-            // will cause mplayer die.
-
-            // Although not support set volume, but at least we support mute.
-            // MPlayer set mute by set volume to zero, we handle it.
-            if (control_vol->left == 0 && control_vol->right == 0)
-                d->muted = true;
-            else
-                d->muted = false;
-            return CONTROL_TRUE;
-        }
-
-        vol = (control_vol->left + control_vol->right) / 200.0;
-        err = AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume,
-                                    kAudioUnitScope_Global, 0, vol, 0);
-
+    }
+    case AOCONTROL_SET_VOLUME: {
+        ao_control_vol_t *control_vol = (ao_control_vol_t *)arg;
+        float vol = (control_vol->left + control_vol->right) / 200.0;
+        OSStatus err =
+            AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume,
+                                  kAudioUnitScope_Global, 0, vol, 0);
         CHECK_CA_ERROR("could not set HAL output volume");
         return CONTROL_TRUE;
-
+    }
     } // end switch
     return CONTROL_UNKNOWN;
 
@@ -202,136 +98,30 @@ coreaudio_error:
     return CONTROL_ERROR;
 }
 
-static void print_list(struct ao *ao)
-{
-    char *help = talloc_strdup(NULL, "Available output devices:\n");
-
-    AudioDeviceID *devs;
-    size_t n_devs;
-
-    OSStatus err =
-        CA_GET_ARY(kAudioObjectSystemObject, kAudioHardwarePropertyDevices,
-                   &devs, &n_devs);
-
-    CHECK_CA_ERROR("Failed to get list of output devices.");
-
-    for (int i = 0; i < n_devs; i++) {
-        char *name;
-        err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &name);
-
-        if (err == noErr)
-            talloc_steal(devs, name);
-        else
-            name = "Unknown";
-
-        help = talloc_asprintf_append(
-                help, "  * %s (id: %" PRIu32 ")\n", name, devs[i]);
-    }
-
-    talloc_free(devs);
-
-coreaudio_error:
-    MP_INFO(ao, "%s", help);
-    talloc_free(help);
-}
-
+static bool init_chmap(struct ao *ao);
 static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd);
-static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd);
 
 static int init(struct ao *ao)
 {
-    OSStatus err;
-    struct priv *p   = ao->priv;
-
-    if (p->opt_list) print_list(ao);
-
-    struct priv_d *d = talloc_zero(p, struct priv_d);
-
-    *d = (struct priv_d) {
-        .muted = false,
-        .stream_asbd_changed = 0,
-        .hog_pid = -1,
-        .stream = 0,
-        .stream_idx = -1,
-        .changed_mixing = false,
-    };
+    struct priv *p = ao->priv;
 
-    p->digital = d;
+    if (p->opt_list) ca_print_device_list(ao);
 
     ao->per_application_mixer = true;
     ao->no_persistent_volume  = true;
 
-    AudioDeviceID selected_device = 0;
-    if (p->opt_device_id < 0) {
-        // device not set by user, get the default one
-        err = CA_GET(kAudioObjectSystemObject,
-                     kAudioHardwarePropertyDefaultOutputDevice,
-                     &selected_device);
-        CHECK_CA_ERROR("could not get default audio device");
-    } else {
-        selected_device = p->opt_device_id;
-    }
-
-    if (mp_msg_test(ao->log, MSGL_V)) {
-        char *name;
-        err = CA_GET_STR(selected_device, kAudioObjectPropertyName, &name);
-        CHECK_CA_ERROR("could not get selected audio device name");
+    OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device);
+    CHECK_CA_ERROR("failed to select device");
 
-        MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
-                       name, selected_device);
-
-        talloc_free(name);
-    }
-
-    // Save selected device id
-    p->device = selected_device;
+    if (!init_chmap(ao))
+        goto coreaudio_error;
 
     ao->format = af_fmt_from_planar(ao->format);
 
-    bool supports_digital = false;
-    /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
-    if (AF_FORMAT_IS_AC3(ao->format)) {
-        if (ca_device_supports_digital(ao, selected_device))
-            supports_digital = true;
-    }
-
-    if (!supports_digital) {
-        AudioChannelLayout *layouts;
-        size_t n_layouts;
-        err = CA_GET_ARY_O(selected_device,
-                           kAudioDevicePropertyPreferredChannelLayout,
-                           &layouts, &n_layouts);
-        CHECK_CA_ERROR("could not get audio device prefered layouts");
-
-        struct mp_chmap_sel chmap_sel = {0};
-        for (int i = 0; i < n_layouts; i++) {
-            struct mp_chmap chmap = {0};
-            if (ca_layout_to_mp_chmap(ao, &layouts[i], &chmap))
-                mp_chmap_sel_add_map(&chmap_sel, &chmap);
-        }
-
-        talloc_free(layouts);
-
-        if (ao->channels.num < 3) {
-            struct mp_chmap chmap;
-            mp_chmap_from_channels(&chmap, ao->channels.num);
-            mp_chmap_sel_add_map(&chmap_sel, &chmap);
-        }
-
-        if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) {
-            MP_ERR(ao, "could not select a suitable channel map among the "
-                       "hardware supported ones. Make sure to configure your "
-                       "output device correctly in 'Audio MIDI Setup.app'\n");
-            goto coreaudio_error;
-        }
-
-    } // closes if (!supports_digital)
-
     // Build ASBD for the input format
     AudioStreamBasicDescription asbd;
     asbd.mSampleRate       = ao->samplerate;
-    asbd.mFormatID         = supports_digital ?
-                             kAudioFormat60958AC3 : kAudioFormatLinearPCM;
+    asbd.mFormatID         = kAudioFormatLinearPCM;
     asbd.mChannelsPerFrame = ao->channels.num;
     asbd.mBitsPerChannel   = af_fmt2bits(ao->format);
     asbd.mFormatFlags      = kAudioFormatFlagIsPacked;
@@ -352,15 +142,52 @@ static int init(struct ao *ao)
 
     ca_print_asbd(ao, "source format:", &asbd);
 
-    if (supports_digital)
-        return init_digital(ao, asbd);
-    else
-        return init_lpcm(ao, asbd);
+    return init_lpcm(ao, asbd);
 
 coreaudio_error:
     return CONTROL_ERROR;
 }
 
+static bool init_chmap(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    OSStatus err;
+    AudioChannelLayout *layouts;
+    size_t n_layouts;
+
+    err = CA_GET_ARY_O(p->device,
+                       kAudioDevicePropertyPreferredChannelLayout,
+                       &layouts, &n_layouts);
+    CHECK_CA_ERROR("could not get audio device prefered layouts");
+
+    struct mp_chmap_sel chmap_sel = {0};
+    for (int i = 0; i < n_layouts; i++) {
+        struct mp_chmap chmap = {0};
+        if (ca_layout_to_mp_chmap(ao, &layouts[i], &chmap))
+            mp_chmap_sel_add_map(&chmap_sel, &chmap);
+    }
+
+    talloc_free(layouts);
+
+    if (ao->channels.num < 3) {
+        struct mp_chmap chmap;
+        mp_chmap_from_channels(&chmap, ao->channels.num);
+        mp_chmap_sel_add_map(&chmap_sel, &chmap);
+    }
+
+    if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels)) {
+        MP_ERR(ao, "could not select a suitable channel map among the "
+                   "hardware supported ones. Make sure to configure your "
+                   "output device correctly in 'Audio MIDI Setup.app'\n");
+        goto coreaudio_error;
+    }
+
+    return true;
+
+coreaudio_error:
+    return false;
+}
+
 static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
 {
     OSStatus err;
@@ -408,8 +235,6 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
                      "can't link audio unit to selected device");
 
     p->buffer = mp_ring_new(p, get_ring_size(ao));
-    print_buffer(ao, p->buffer);
-
     AURenderCallbackStruct render_cb = (AURenderCallbackStruct) {
         .inputProc       = render_cb_lpcm,
         .inputProcRefCon = ao,
@@ -434,153 +259,11 @@ coreaudio_error:
     return CONTROL_ERROR;
 }
 
-static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
-{
-    struct priv *p = ao->priv;
-    struct priv_d *d = p->digital;
-    OSStatus err = noErr;
-
-    uint32_t is_alive = 1;
-    err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
-    CHECK_CA_WARN("could not check whether device is alive");
-
-    if (!is_alive)
-        MP_WARN(ao , "device is not alive\n");
-
-    p->is_digital = 1;
-
-    err = ca_lock_device(p->device, &d->hog_pid);
-    CHECK_CA_WARN("failed to set hogmode");
-
-    err = ca_disable_mixing(ao, p->device, &d->changed_mixing);
-    CHECK_CA_WARN("failed to disable mixing");
-
-    AudioStreamID *streams;
-    size_t n_streams;
-
-    /* Get a list of all the streams on this device. */
-    err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
-                       &streams, &n_streams);
-
-    CHECK_CA_ERROR("could not get number of streams");
-
-    for (int i = 0; i < n_streams && d->stream_idx < 0; i++) {
-        bool digital = ca_stream_supports_digital(ao, streams[i]);
-
-        if (digital) {
-            err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
-                         &d->original_asbd);
-            if (!CHECK_CA_WARN("could not get stream's physical format to "
-                               "revert to, getting the next one"))
-                continue;
-
-            AudioStreamRangedDescription *formats;
-            size_t n_formats;
-
-            err = CA_GET_ARY(streams[i],
-                             kAudioStreamPropertyAvailablePhysicalFormats,
-                             &formats, &n_formats);
-
-            if (!CHECK_CA_WARN("could not get number of stream formats"))
-                continue; // try next one
-
-            int req_rate_format = -1;
-            int max_rate_format = -1;
-
-            d->stream = streams[i];
-            d->stream_idx = i;
-
-            for (int j = 0; j < n_formats; j++)
-                if (ca_format_is_digital(formats[j].mFormat)) {
-                    // select the digital format that has exactly the same
-                    // samplerate. If an exact match cannot be found, select
-                    // the format with highest samplerate as backup.
-                    if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) {
-                        req_rate_format = j;
-                        break;
-                    } else if (max_rate_format < 0 ||
-                        formats[j].mFormat.mSampleRate >
-                        formats[max_rate_format].mFormat.mSampleRate)
-                        max_rate_format = j;
-                }
-
-            if (req_rate_format >= 0)
-                d->stream_asbd = formats[req_rate_format].mFormat;
-            else
-                d->stream_asbd = formats[max_rate_format].mFormat;
-
-            talloc_free(formats);
-        }
-    }
-
-    talloc_free(streams);
-
-    if (d->stream_idx < 0) {
-        MP_WARN(ao , "can't find any digital output stream format\n");
-        goto coreaudio_error;
-    }
-
-    if (!ca_change_format(ao, d->stream, d->stream_asbd))
-        goto coreaudio_error;
-
-    void *changed = (void *) &(d->stream_asbd_changed);
-    err = ca_enable_device_listener(p->device, changed);
-    CHECK_CA_ERROR("cannot install format change listener during init");
-
-#if BYTE_ORDER == BIG_ENDIAN
-    if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
-#else
-    /* tell mplayer that we need a byteswap on AC3 streams, */
-    if (d->stream_asbd.mFormatID & kAudioFormat60958AC3)
-        ao->format = AF_FORMAT_AC3_LE;
-    else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
-#endif
-        MP_WARN(ao, "stream has non-native byte order, output may fail\n");
-
-    ao->samplerate = d->stream_asbd.mSampleRate;
-    ao->bps = ao->samplerate *
-                  (d->stream_asbd.mBytesPerPacket /
-                   d->stream_asbd.mFramesPerPacket);
-
-    p->buffer = mp_ring_new(p, get_ring_size(ao));
-    print_buffer(ao, p->buffer);
-
-    err = AudioDeviceCreateIOProcID(p->device,
-                                    (AudioDeviceIOProc)render_cb_digital,
-                                    (void *)ao,
-                                    &d->render_cb);
-
-    CHECK_CA_ERROR("failed to register digital render callback");
-
-    reset(ao);
-
-    return CONTROL_TRUE;
-
-coreaudio_error:
-    err = ca_unlock_device(p->device, &d->hog_pid);
-    CHECK_CA_WARN("can't release hog mode");
-    return CONTROL_ERROR;
-}
-
 static int play(struct ao *ao, void **data, int samples, int flags)
 {
-    struct priv *p   = ao->priv;
-    struct priv_d *d = p->digital;
+    struct priv *p       = ao->priv;
     void *output_samples = data[0];
-    int num_bytes = samples * ao->sstride;
-
-    // Check whether we need to reset the digital output stream.
-    if (p->is_digital && d->stream_asbd_changed) {
-        d->stream_asbd_changed = 0;
-        if (ca_stream_supports_digital(ao, d->stream)) {
-            if (!ca_change_format(ao, d->stream, d->stream_asbd)) {
-                MP_WARN(ao , "can't restore digital output\n");
-            } else {
-                MP_WARN(ao, "restoring digital output succeeded.\n");
-                reset(ao);
-            }
-        }
-    }
+    int num_bytes        = samples * ao->sstride;
 
     int wrote = mp_ring_write(p->buffer, output_samples, num_bytes);
     audio_resume(ao);
@@ -612,52 +295,20 @@ static float get_delay(struct ao *ao)
 static void uninit(struct ao *ao)
 {
     struct priv *p = ao->priv;
-    OSStatus err = noErr;
-
-    if (!p->is_digital) {
-        AudioOutputUnitStop(p->audio_unit);
-        AudioUnitUninitialize(p->audio_unit);
-        AudioComponentInstanceDispose(p->audio_unit);
-    } else {
-        struct priv_d *d = p->digital;
-
-        void *changed = (void *) &(d->stream_asbd_changed);
-        err = ca_disable_device_listener(p->device, changed);
-        CHECK_CA_WARN("can't remove device listener, this may cause a crash");
-
-        err = AudioDeviceStop(p->device, d->render_cb);
-        CHECK_CA_WARN("failed to stop audio device");
-
-        err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
-        CHECK_CA_WARN("failed to remove device render callback");
-
-        if (!ca_change_format(ao, d->stream, d->original_asbd))
-            MP_WARN(ao, "can't revert to original device format");
-
-        err = ca_enable_mixing(ao, p->device, d->changed_mixing);
-        CHECK_CA_WARN("can't re-enable mixing");
-
-        err = ca_unlock_device(p->device, &d->hog_pid);
-        CHECK_CA_WARN("can't release hog mode");
-    }
+    AudioOutputUnitStop(p->audio_unit);
+    AudioUnitUninitialize(p->audio_unit);
+    AudioComponentInstanceDispose(p->audio_unit);
 }
 
 static void audio_pause(struct ao *ao)
 {
     struct priv *p = ao->priv;
-    OSErr err = noErr;
 
     if (p->paused)
         return;
 
-    if (!p->is_digital) {
-        err = AudioOutputUnitStop(p->audio_unit);
-        CHECK_CA_WARN("can't stop audio unit");
-    } else {
-        struct priv_d *d = p->digital;
-        err = AudioDeviceStop(p->device, d->render_cb);
-        CHECK_CA_WARN("can't stop digital device");
-    }
+    OSStatus err = AudioOutputUnitStop(p->audio_unit);
+    CHECK_CA_WARN("can't stop audio unit");
 
     p->paused = true;
 }
@@ -665,19 +316,12 @@ static void audio_pause(struct ao *ao)
 static void audio_resume(struct ao *ao)
 {
     struct priv *p = ao->priv;
-    OSErr err = noErr;
 
     if (!p->paused)
         return;
 
-    if (!p->is_digital) {
-        err = AudioOutputUnitStart(p->audio_unit);
-        CHECK_CA_WARN("can't start audio unit");
-    } else {
-        struct priv_d *d = p->digital;
-        err = AudioDeviceStart(p->device, d->render_cb);
-        CHECK_CA_WARN("can't start digital device");
-    }
+    OSStatus err = AudioOutputUnitStart(p->audio_unit);
+    CHECK_CA_WARN("can't start audio unit");
 
     p->paused = false;
 }
@@ -685,7 +329,7 @@ static void audio_resume(struct ao *ao)
 #define OPT_BASE_STRUCT struct priv
 
 const struct ao_driver audio_out_coreaudio = {
-    .description = "CoreAudio (OS X Audio Output)",
+    .description = "CoreAudio AudioUnit",
     .name      = "coreaudio",
     .uninit    = uninit,
     .init      = init,
diff --git a/audio/out/ao_coreaudio_device.c b/audio/out/ao_coreaudio_device.c
new file mode 100644
index 0000000000..c86ffbd700
--- /dev/null
+++ b/audio/out/ao_coreaudio_device.c
@@ -0,0 +1,449 @@
+/*
+ * CoreAudio audio output driver for Mac OS X
+ *
+ * original copyright (C) Timothy J. Wood - Aug 2000
+ * ported to MPlayer libao2 by Dan Christiansen
+ *
+ * Chris Roccati
+ * Stefano Pigozzi
+ *
+ * The S/PDIF part of the code is based on the auhal audio output
+ * module from VideoLAN:
+ * Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * along with MPlayer; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/*
+ * The MacOS X 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).
+ */
+
+#include "config.h"
+#include "ao.h"
+#include "internal.h"
+#include "audio/format.h"
+#include "osdep/timer.h"
+#include "options/m_option.h"
+#include "misc/ring.h"
+#include "common/msg.h"
+#include "audio/out/ao_coreaudio_properties.h"
+#include "audio/out/ao_coreaudio_utils.h"
+
+static void audio_pause(struct ao *ao);
+static void audio_resume(struct ao *ao);
+static void reset(struct ao *ao);
+
+struct priv {
+    AudioDeviceID device;   // selected device
+
+    bool paused;
+
+    struct mp_ring *buffer;
+
+    // digital render callback
+    AudioDeviceIOProcID render_cb;
+
+    // pid set for hog mode, (-1) means that hog mode on the device was
+    // released. hog mode is exclusive access to a device
+    pid_t hog_pid;
+
+    // stream selected for digital playback by the detection in init
+    AudioStreamID stream;
+
+    // stream index in an AudioBufferList
+    int stream_idx;
+
+    // format we changed the stream to: for the digital case each application
+    // sets the stream format for a device to what it needs
+    AudioStreamBasicDescription stream_asbd;
+    AudioStreamBasicDescription original_asbd;
+
+    bool changed_mixing;
+    int stream_asbd_changed;
+    bool muted;
+
+    // options
+    int opt_device_id;
+    int opt_list;
+};
+
+static int get_ring_size(struct ao *ao)
+{
+    return af_fmt_seconds_to_bytes(
+            ao->format, 0.5, ao->channels.num, ao->samplerate);
+}
+
+static OSStatus render_cb_digital(
+        AudioDeviceID device, const AudioTimeStamp *ts,
+        const void *in_data, const AudioTimeStamp *in_ts,
+        AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
+{
+    struct ao *ao    = ctx;
+    struct priv *p   = ao->priv;
+    AudioBuffer buf  = out_data->mBuffers[p->stream_idx];
+    int requested    = buf.mDataByteSize;
+
+    if (p->muted)
+        mp_ring_drain(p->buffer, requested);
+    else
+        mp_ring_read(p->buffer, buf.mData, requested);
+
+    return noErr;
+}
+
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct priv *p = ao->priv;
+    ao_control_vol_t *control_vol;
+    switch (cmd) {
+    case AOCONTROL_GET_VOLUME:
+        control_vol = (ao_control_vol_t *)arg;
+        // Digital output has no volume adjust.
+        int digitalvol = p->muted ? 0 : 100;
+        *control_vol = (ao_control_vol_t) {
+            .left = digitalvol, .right = digitalvol,
+        };
+        return CONTROL_TRUE;
+
+    case AOCONTROL_SET_VOLUME:
+        control_vol = (ao_control_vol_t *)arg;
+        // Digital output can not set volume. Here we have to return true
+        // to make mixer forget it. Else mixer will add a soft filter,
+        // that's not we expected and the filter not support ac3 stream
+        // will cause mplayer die.
+
+        // Although not support set volume, but at least we support mute.
+        // MPlayer set mute by set volume to zero, we handle it.
+        if (control_vol->left == 0 && control_vol->right == 0)
+            p->muted = true;
+        else
+            p->muted = false;
+        return CONTROL_TRUE;
+
+    } // end switch
+    return CONTROL_UNKNOWN;
+}
+
+static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd);
+
+static int init(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    if (p->opt_list) ca_print_device_list(ao);
+
+    *p = (struct priv) {
+        .muted = false,
+        .stream_asbd_changed = 0,
+        .hog_pid = -1,
+        .stream = 0,
+        .stream_idx = -1,
+        .changed_mixing = false,
+    };
+
+    OSStatus err = ca_select_device(ao, p->opt_device_id, &p->device);
+    CHECK_CA_ERROR("failed to select device");
+
+    ao->format = af_fmt_from_planar(ao->format);
+
+    bool supports_digital = false;
+    /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
+    if (AF_FORMAT_IS_AC3(ao->format)) {
+        if (ca_device_supports_digital(ao, p->device))
+            supports_digital = true;
+    }
+
+    if (!supports_digital) {
+        MP_ERR(ao, "selected device doesn't support digital formats\n");
+        goto coreaudio_error;
+    } // closes if (!supports_digital)
+
+    // Build ASBD for the input format
+    AudioStreamBasicDescription asbd;
+    asbd.mSampleRate       = ao->samplerate;
+    asbd.mFormatID         = kAudioFormat60958AC3;
+    asbd.mChannelsPerFrame = ao->channels.num;
+    asbd.mBitsPerChannel   = af_fmt2bits(ao->format);
+    asbd.mFormatFlags      = kAudioFormatFlagIsPacked;
+
+    if ((ao->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F)
+        asbd.mFormatFlags |= kAudioFormatFlagIsFloat;
+
+    if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI)
+        asbd.mFormatFlags |= kAudioFormatFlagIsSignedInteger;
+
+    if ((ao->format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
+        asbd.mFormatFlags |= kAudioFormatFlagIsBigEndian;
+
+    asbd.mFramesPerPacket = 1;
+    asbd.mBytesPerPacket = asbd.mBytesPerFrame =
+        asbd.mFramesPerPacket * asbd.mChannelsPerFrame *
+        (asbd.mBitsPerChannel / 8);
+
+    ca_print_asbd(ao, "source format:", &asbd);
+
+    return init_digital(ao, asbd);
+
+coreaudio_error:
+    return CONTROL_ERROR;
+}
+
+static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
+{
+    struct priv *p = ao->priv;
+    OSStatus err = noErr;
+
+    uint32_t is_alive = 1;
+    err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
+    CHECK_CA_WARN("could not check whether device is alive");
+
+    if (!is_alive)
+        MP_WARN(ao , "device is not alive\n");
+
+    err = ca_lock_device(p->device, &p->hog_pid);
+    CHECK_CA_WARN("failed to set hogmode");
+
+    err = ca_disable_mixing(ao, p->device, &p->changed_mixing);
+    CHECK_CA_WARN("failed to disable mixing");
+
+    AudioStreamID *streams;
+    size_t n_streams;
+
+    /* Get a list of all the streams on this device. */
+    err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
+                       &streams, &n_streams);
+
+    CHECK_CA_ERROR("could not get number of streams");
+
+    for (int i = 0; i < n_streams && p->stream_idx < 0; i++) {
+        bool digital = ca_stream_supports_digital(ao, streams[i]);
+
+        if (digital) {
+            err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
+                         &p->original_asbd);
+            if (!CHECK_CA_WARN("could not get stream's physical format to "
+                               "revert to, getting the next one"))
+                continue;
+
+            AudioStreamRangedDescription *formats;
+            size_t n_formats;
+
+            err = CA_GET_ARY(streams[i],
+                             kAudioStreamPropertyAvailablePhysicalFormats,
+                             &formats, &n_formats);
+
+            if (!CHECK_CA_WARN("could not get number of stream formats"))
+                continue; // try next one
+
+            int req_rate_format = -1;
+            int max_rate_format = -1;
+
+            p->stream = streams[i];
+            p->stream_idx = i;
+
+            for (int j = 0; j < n_formats; j++)
+                if (ca_format_is_digital(formats[j].mFormat)) {
+                    // select the digital format that has exactly the same
+                    // samplerate. If an exact match cannot be found, select
+                    // the format with highest samplerate as backup.
+                    if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) {
+                        req_rate_format = j;
+                        break;
+                    } else if (max_rate_format < 0 ||
+                        formats[j].mFormat.mSampleRate >
+                        formats[max_rate_format].mFormat.mSampleRate)
+                        max_rate_format = j;
+                }
+
+            if (req_rate_format >= 0)
+                p->stream_asbd = formats[req_rate_format].mFormat;
+            else
+                p->stream_asbd = formats[max_rate_format].mFormat;
+
+            talloc_free(formats);
+        }
+    }
+
+    talloc_free(streams);
+
+    if (p->strea