From e37634e908c7563e6281997d50800a2f9633d8d2 Mon Sep 17 00:00:00 2001
From: Chris Robinson <chris.kcat@gmail.com>
Date: Fri, 23 Mar 2018 11:58:11 -0700
Subject: [PATCH] Remove an unused struct and move some functions to where
 they're used

---
 Alc/backends/coreaudio.c | 62 +++++++++++++++++++-----------------------------
 1 file changed, 24 insertions(+), 38 deletions(-)

diff --git a/Alc/backends/coreaudio.c b/Alc/backends/coreaudio.c
index d6d32cc8..eeb19dab 100644
--- a/Alc/backends/coreaudio.c
+++ b/Alc/backends/coreaudio.c
@@ -36,45 +36,9 @@
 #include "backends/base.h"
 
 
-typedef struct {
-    AudioUnit audioUnit;
-
-    ALuint frameSize;
-    ALdouble sampleRateRatio;              // Ratio of hardware sample rate / requested sample rate
-    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
-
-    AudioConverterRef audioConverter;      // Sample rate converter if needed
-    AudioBufferList *bufferList;           // Buffer for data coming from the input device
-    ALCvoid *resampleBuffer;               // Buffer for returned RingBuffer data when resampling
-
-    ll_ringbuffer_t *ring;
-} ca_data;
-
 static const ALCchar ca_device[] = "CoreAudio Default";
 
 
-static AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
-{
-    AudioBufferList *list;
-
-    list = calloc(1, FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize);
-    if(list)
-    {
-        list->mNumberBuffers = 1;
-
-        list->mBuffers[0].mNumberChannels = channelCount;
-        list->mBuffers[0].mDataByteSize = byteSize;
-        list->mBuffers[0].mData = &list->mBuffers[1];
-    }
-    return list;
-}
-
-static void destroy_buffer_list(AudioBufferList *list)
-{
-    free(list);
-}
-
-
 typedef struct ALCcoreAudioPlayback {
     DERIVE_FROM_TYPE(ALCbackend);
 
@@ -125,10 +89,10 @@ static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
     ALCcoreAudioPlayback *self = inRefCon;
     ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
 
-    ALCdevice_Lock(device);
+    ALCcoreAudioPlayback_lock(self);
     aluMixData(device, ioData->mBuffers[0].mData,
                ioData->mBuffers[0].mDataByteSize / self->frameSize);
-    ALCdevice_Unlock(device);
+    ALCcoreAudioPlayback_unlock(self);
 
     return noErr;
 }
@@ -381,6 +345,28 @@ DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture)
 DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture);
 
 
+static AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
+{
+    AudioBufferList *list;
+
+    list = calloc(1, FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize);
+    if(list)
+    {
+        list->mNumberBuffers = 1;
+
+        list->mBuffers[0].mNumberChannels = channelCount;
+        list->mBuffers[0].mDataByteSize = byteSize;
+        list->mBuffers[0].mData = &list->mBuffers[1];
+    }
+    return list;
+}
+
+static void destroy_buffer_list(AudioBufferList *list)
+{
+    free(list);
+}
+
+
 static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device)
 {
     ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-- 
2.11.4.GIT