osxaudio: Attempt to configure the segment size in CoreAudio

Set the BufferFrame size in CoreAudio so it will deliver data
in ringbuffer segment units when recording. Otherwise, CoreAudio
will provide data in whatever granularity it wants, with no
relationship to the requested latency-time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>

subprojects/gst-plugins-good/sys/osxaudio/gstosxaudioringbuffer.c

@@ -174,7 +174,8 @@ gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
 {
   gboolean ret = FALSE, is_passthrough = FALSE;
   GstOsxAudioRingBuffer *osxbuf;
-  AudioStreamBasicDescription format;
+  AudioStreamBasicDescription format = { 0 };
+  guint32 frames_per_packet = 0;
 
   osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
 
@@ -225,6 +226,8 @@ gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
         (spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
         G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
     spec->segtotal = spec->buffer_time / spec->latency_time;
+    frames_per_packet = spec->segsize / GST_AUDIO_INFO_BPF (&spec->info);
+
     is_passthrough = FALSE;
   }
 
@@ -239,7 +242,7 @@ gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
   buf->memory = g_malloc0 (buf->size);
 
   ret = gst_core_audio_initialize (osxbuf->core_audio, format, spec->caps,
-      is_passthrough);
+      frames_per_packet, is_passthrough);
 
   if (!ret) {
     g_free (buf->memory);

subprojects/gst-plugins-good/sys/osxaudio/gstosxcoreaudio.c

@@ -208,22 +208,26 @@ gst_core_audio_get_samples_and_latency (GstCoreAudio * core_audio,
 
 gboolean
 gst_core_audio_initialize (GstCoreAudio * core_audio,
-    AudioStreamBasicDescription format, GstCaps * caps, gboolean is_passthrough)
+    AudioStreamBasicDescription format, GstCaps * caps,
+    guint32 frames_per_packet, gboolean is_passthrough)
 {
-  guint32 frame_size;
-
   GST_DEBUG_OBJECT (core_audio,
       "Initializing: passthrough:%d caps:%" GST_PTR_FORMAT, is_passthrough,
       caps);
 
   if (!gst_core_audio_initialize_impl (core_audio, format, caps,
-          is_passthrough, &frame_size)) {
+          is_passthrough, &frames_per_packet)) {
     return FALSE;
   }
 
   if (core_audio->is_src) {
     /* create AudioBufferList needed for recording */
-    core_audio->recBufferSize = frame_size * format.mBytesPerFrame;
+    core_audio->recBufferSize = frames_per_packet * format.mBytesPerFrame;
+
+    GST_DEBUG_OBJECT (core_audio,
+        "Allocating record buffers %u bytes %u frames",
+        core_audio->recBufferSize, frames_per_packet);
+
     core_audio->recBufferList =
         buffer_list_alloc (format.mChannelsPerFrame, core_audio->recBufferSize,
         /* Currently always TRUE (i.e. interleaved) */

subprojects/gst-plugins-good/sys/osxaudio/gstosxcoreaudio.h

@@ -127,6 +127,7 @@ gboolean gst_core_audio_close                                (GstCoreAudio *core
 gboolean gst_core_audio_initialize                           (GstCoreAudio *core_audio,
                                                               AudioStreamBasicDescription format,
                                                               GstCaps *caps,
+                                                              guint32 frames_per_packet,
                                                               gboolean is_passthrough);
 
 void gst_core_audio_uninitialize                             (GstCoreAudio *core_audio);

subprojects/gst-plugins-good/sys/osxaudio/gstosxcoreaudiohal.c

@@ -1174,6 +1174,24 @@ gst_core_audio_initialize_impl (GstCoreAudio * core_audio,
 
     if (core_audio->is_src) {
       propertySize = sizeof (*frame_size);
+
+      // Attempt to configure the requested frame size if smaller than the device's
+      // This will apparently modify the size for all audio devices in the current process so should
+      // be done conservatively
+      if (frame_size != 0) {
+        guint32 cur_frame_size;
+        status = AudioUnitGetProperty (core_audio->audiounit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0,   /* N/A for global */
+            &cur_frame_size, &propertySize);
+        if (!status && *frame_size < cur_frame_size) {
+          status = AudioUnitSetProperty (core_audio->audiounit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, /* N/A for global */
+              frame_size, propertySize);
+          if (status) {
+            GST_WARNING_OBJECT (core_audio->osxbuf,
+                "Failed to set desired frame size of %u: %d", *frame_size,
+                (int) status);
+          }
+        }
+      }
       status = AudioUnitGetProperty (core_audio->audiounit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0,     /* N/A for global */
           frame_size, &propertySize);