ao_coreaudio_exclusive: port to pull API, fix latency calculations

Instead of maintaining a private ring buffer, use the generic support
for audio APIs with pull callbacks (internally called AO pull API). This
also fixes latency calculations: instead of just returning the
ringbuffer status, the audio playback state is calculated better and
includes interpolation.

The main reason this wasn't done earlier was mid-stream format
switching. The pull API can now handle it (in a way) by destroying and
recreating the AO. This is a bit brutal, but quite simple. It's untested
in this new AO, though. Some details might not be right, like how ot
restores the old format when reloading.

audio/out/ao_coreaudio_exclusive.c

@@ -35,21 +35,18 @@
  * when you are wanting to do good buffering of audio).
  */
 
+#include <CoreAudio/HostTime.h>
+
 #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);
-
 static bool ca_format_is_digital(AudioStreamBasicDescription asbd)
 {
     switch (asbd.mFormatID)
@@ -310,8 +307,6 @@ struct priv {
 
     bool paused;
 
-    struct mp_ring *buffer;
-
     // digital render callback
     AudioDeviceIOProcID render_cb;
 
@@ -332,13 +327,10 @@ struct priv {
 
     bool changed_mixing;
     int stream_asbd_changed;
-};
+    bool reload_requested;
 
-static int get_ring_size(struct ao *ao)
+    uint32_t hw_latency_us;
-{
+};
-    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,
@@ -350,7 +342,29 @@ static OSStatus render_cb_digital(
     AudioBuffer buf  = out_data->mBuffers[p->stream_idx];
     int requested    = buf.mDataByteSize;
 
-    mp_ring_read(p->buffer, buf.mData, requested);
+    int pseudo_frames = requested / ao->sstride;
+
+    // we expect the callback to read full frames, which are aligned accordingly
+    if (pseudo_frames * ao->sstride != requested) {
+        MP_ERR(ao, "Unsupported unaligned read of %d bytes.\n", requested);
+        return kAudioHardwareUnspecifiedError;
+    }
+
+    int64_t end = mp_time_us();
+    end += p->hw_latency_us + ca_get_latency(ts)
+        + ca_frames_to_us(ao, pseudo_frames);
+
+    ao_read_data(ao, &buf.mData, pseudo_frames, end);
+
+    // Check whether we need to reset the digital output stream.
+    if (p->stream_asbd_changed) {
+        p->stream_asbd_changed = 0;
+        if (!p->reload_requested && ca_stream_supports_digital(ao, p->stream)) {
+            p->reload_requested = true;
+            ao_request_reload(ao);
+            MP_INFO(ao, "Stream format changed! Reloading.\n");
+        }
+    }
 
     return noErr;
 }
@@ -487,17 +501,22 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
                   (p->stream_asbd.mBytesPerPacket /
                    p->stream_asbd.mFramesPerPacket);
 
-    p->buffer = mp_ring_new(p, get_ring_size(ao));
+    uint32_t latency_frames = 0;
+    err = CA_GET_O(p->device, kAudioDevicePropertyLatency, &latency_frames);
+    if (err != noErr) {
+        CHECK_CA_WARN("cannot get device latency");
+        latency_frames = 0;
+    }
+
+    p->hw_latency_us = ca_frames_to_us(ao, latency_frames);
+    MP_VERBOSE(ao, "base latency: %d microseconds\n", (int)p->hw_latency_us);
 
     err = AudioDeviceCreateIOProcID(p->device,
                                     (AudioDeviceIOProc)render_cb_digital,
                                     (void *)ao,
                                     &p->render_cb);
-
     CHECK_CA_ERROR("failed to register digital render callback");
 
-    reset(ao);
-
     return CONTROL_TRUE;
 
 coreaudio_error:
@@ -506,52 +525,6 @@ coreaudio_error:
     return CONTROL_ERROR;
 }
 
-static int play(struct ao *ao, void **data, int samples, int flags)
-{
-    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->stream_asbd_changed) {
-        p->stream_asbd_changed = 0;
-        if (ca_stream_supports_digital(ao, p->stream)) {
-            if (!ca_change_format(ao, p->stream, p->stream_asbd)) {
-                MP_WARN(ao , "can't restore digital output\n");
-            } else {
-                MP_WARN(ao, "restoring digital output succeeded.\n");
-                reset(ao);
-            }
-        }
-    }
-
-    int wrote = mp_ring_write(p->buffer, output_samples, num_bytes);
-    audio_resume(ao);
-
-    return wrote / ao->sstride;
-}
-
-static void reset(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    audio_pause(ao);
-    mp_ring_reset(p->buffer);
-}
-
-static int get_space(struct ao *ao)
-{
-    struct priv *p = ao->priv;
-    return mp_ring_available(p->buffer) / ao->sstride;
-}
-
-static double get_delay(struct ao *ao)
-{
-    // FIXME: should also report the delay of coreaudio itself (hardware +
-    // internal buffers)
-    struct priv *p = ao->priv;
-    return mp_ring_buffered(p->buffer) / (double)ao->bps;
-}
-
 static void uninit(struct ao *ao)
 {
     struct priv *p = ao->priv;
@@ -581,26 +554,16 @@ static void audio_pause(struct ao *ao)
 {
     struct priv *p = ao->priv;
 
-    if (p->paused)
-        return;
-
     OSStatus err = AudioDeviceStop(p->device, p->render_cb);
     CHECK_CA_WARN("can't stop digital device");
-
-    p->paused = true;
 }
 
 static void audio_resume(struct ao *ao)
 {
     struct priv *p = ao->priv;
 
-    if (!p->paused)
-        return;
-
     OSStatus err = AudioDeviceStart(p->device, p->render_cb);
     CHECK_CA_WARN("can't start digital device");
-
-    p->paused = false;
 }
 
 #define OPT_BASE_STRUCT struct priv
@@ -610,10 +573,6 @@ const struct ao_driver audio_out_coreaudio_exclusive = {
     .name      = "coreaudio_exclusive",
     .uninit    = uninit,
     .init      = init,
-    .play      = play,
-    .get_space = get_space,
-    .get_delay = get_delay,
-    .reset     = reset,
     .pause     = audio_pause,
     .resume    = audio_resume,
     .list_devs = ca_get_device_list,