1 files changed, 345 insertions, 241 deletions

diff --git a/Alc/backends/coreaudio.c b/Alc/backends/coreaudio.c
index 43e881da..adb01fa6 100644
--- a/Alc/backends/coreaudio.c
+++ b/Alc/backends/coreaudio.c
@@ -23,195 +23,145 @@
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
-#include <alloca.h>
 
 #include "alMain.h"
 #include "alu.h"
+#include "ringbuffer.h"
 
-#include <CoreServices/CoreServices.h>
 #include <unistd.h>
 #include <AudioUnit/AudioUnit.h>
 #include <AudioToolbox/AudioToolbox.h>
 
+#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
+static const ALCchar ca_device[] = "CoreAudio Default";
 
-    RingBuffer *ring;
-} ca_data;
 
-static const ALCchar ca_device[] = "CoreAudio Default";
+typedef struct ALCcoreAudioPlayback {
+    DERIVE_FROM_TYPE(ALCbackend);
 
+    AudioUnit audioUnit;
 
-static void destroy_buffer_list(AudioBufferList* list)
-{
-    if(list)
-    {
-        UInt32 i;
-        for(i = 0;i < list->mNumberBuffers;i++)
-            free(list->mBuffers[i].mData);
-        free(list);
-    }
-}
+    ALuint frameSize;
+    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
+} ALCcoreAudioPlayback;
 
-static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
-{
-    AudioBufferList *list;
+static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device);
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self);
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name);
+static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self);
+static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self);
+static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self);
+static DECLARE_FORWARD2(ALCcoreAudioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ALCuint, availableSamples)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, unlock)
+DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioPlayback)
 
-    list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
-    if(list)
-    {
-        list->mNumberBuffers = 1;
+DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback);
 
-        list->mBuffers[0].mNumberChannels = channelCount;
-        list->mBuffers[0].mDataByteSize = byteSize;
-        list->mBuffers[0].mData = malloc(byteSize);
-        if(list->mBuffers[0].mData == NULL)
-        {
-            free(list);
-            list = NULL;
-        }
-    }
-    return list;
-}
 
-static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
-                            UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
+static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device)
 {
-    ALCdevice *device = (ALCdevice*)inRefCon;
-    ca_data *data = (ca_data*)device->ExtraData;
-
-    aluMixData(device, ioData->mBuffers[0].mData,
-               ioData->mBuffers[0].mDataByteSize / data->frameSize);
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self);
 
-    return noErr;
+    self->frameSize = 0;
+    memset(&self->format, 0, sizeof(self->format));
 }
 
-static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
-        AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
 {
-    ALCdevice *device = (ALCdevice*)inUserData;
-    ca_data *data = (ca_data*)device->ExtraData;
+    AudioUnitUninitialize(self->audioUnit);
+    AudioComponentInstanceDispose(self->audioUnit);
 
-    // Read from the ring buffer and store temporarily in a large buffer
-    ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets));
-
-    // Set the input data
-    ioData->mNumberBuffers = 1;
-    ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
-    ioData->mBuffers[0].mData = data->resampleBuffer;
-    ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
-
-    return noErr;
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
 }
 
-static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
-                                    const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
-                                    UInt32 inNumberFrames, AudioBufferList *ioData)
-{
-    ALCdevice *device = (ALCdevice*)inRefCon;
-    ca_data *data = (ca_data*)device->ExtraData;
-    AudioUnitRenderActionFlags flags = 0;
-    OSStatus err;
 
-    // fill the bufferList with data from the input device
-    err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
-    if(err != noErr)
-    {
-        ERR("AudioUnitRender error: %d\n", err);
-        return err;
-    }
+static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
+  AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp),
+  UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData)
+{
+    ALCcoreAudioPlayback *self = inRefCon;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
 
-    WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
+    ALCcoreAudioPlayback_lock(self);
+    aluMixData(device, ioData->mBuffers[0].mData,
+               ioData->mBuffers[0].mDataByteSize / self->frameSize);
+    ALCcoreAudioPlayback_unlock(self);
 
     return noErr;
 }
 
-static ALCenum ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
+
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
 {
-    ComponentDescription desc;
-    Component comp;
-    ca_data *data;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioComponentDescription desc;
+    AudioComponent comp;
     OSStatus err;
 
-    if(!deviceName)
-        deviceName = ca_device;
-    else if(strcmp(deviceName, ca_device) != 0)
+    if(!name)
+        name = ca_device;
+    else if(strcmp(name, ca_device) != 0)
         return ALC_INVALID_VALUE;
 
     /* open the default output unit */
     desc.componentType = kAudioUnitType_Output;
+#if TARGET_OS_IOS
+    desc.componentSubType = kAudioUnitSubType_RemoteIO;
+#else
     desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+#endif
     desc.componentManufacturer = kAudioUnitManufacturer_Apple;
     desc.componentFlags = 0;
     desc.componentFlagsMask = 0;
 
-    comp = FindNextComponent(NULL, &desc);
+    comp = AudioComponentFindNext(NULL, &desc);
     if(comp == NULL)
     {
-        ERR("FindNextComponent failed\n");
+        ERR("AudioComponentFindNext failed\n");
         return ALC_INVALID_VALUE;
     }
 
-    data = calloc(1, sizeof(*data));
-
-    err = OpenAComponent(comp, &data->audioUnit);
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
     if(err != noErr)
     {
-        ERR("OpenAComponent failed\n");
-        free(data);
+        ERR("AudioComponentInstanceNew failed\n");
         return ALC_INVALID_VALUE;
     }
 
     /* init and start the default audio unit... */
-    err = AudioUnitInitialize(data->audioUnit);
+    err = AudioUnitInitialize(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioUnitInitialize failed\n");
-        CloseComponent(data->audioUnit);
-        free(data);
+        AudioComponentInstanceDispose(self->audioUnit);
         return ALC_INVALID_VALUE;
     }
 
-    al_string_copy_cstr(&device->DeviceName, deviceName);
-    device->ExtraData = data;
+    alstr_copy_cstr(&device->DeviceName, name);
     return ALC_NO_ERROR;
 }
 
-static void ca_close_playback(ALCdevice *device)
+static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-
-    AudioUnitUninitialize(data->audioUnit);
-    CloseComponent(data->audioUnit);
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static ALCboolean ca_reset_playback(ALCdevice *device)
-{
-    ca_data *data = (ca_data*)device->ExtraData;
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
     AudioStreamBasicDescription streamFormat;
     AURenderCallbackStruct input;
     OSStatus err;
     UInt32 size;
 
-    err = AudioUnitUninitialize(data->audioUnit);
+    err = AudioUnitUninitialize(self->audioUnit);
     if(err != noErr)
         ERR("-- AudioUnitUninitialize failed.\n");
 
     /* retrieve default output unit's properties (output side) */
     size = sizeof(AudioStreamBasicDescription);
-    err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
     if(err != noErr || size != sizeof(AudioStreamBasicDescription))
     {
         ERR("AudioUnitGetProperty failed\n");
@@ -229,7 +179,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
 #endif
 
     /* set default output unit's input side to match output side */
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -238,7 +188,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
 
     if(device->Frequency != streamFormat.mSampleRate)
     {
-        device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
+        device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates *
                                       streamFormat.mSampleRate /
                                       device->Frequency);
         device->Frequency = streamFormat.mSampleRate;
@@ -313,7 +263,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
                                  kLinearPCMFormatFlagIsPacked;
 
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -321,11 +271,11 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     }
 
     /* setup callback */
-    data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
-    input.inputProc = ca_callback;
-    input.inputProcRefCon = device;
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    input.inputProc = ALCcoreAudioPlayback_MixerProc;
+    input.inputProcRefCon = self;
 
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -333,7 +283,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     }
 
     /* init the default audio unit... */
-    err = AudioUnitInitialize(data->audioUnit);
+    err = AudioUnitInitialize(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioUnitInitialize failed\n");
@@ -343,12 +293,9 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static ALCboolean ca_start_playback(ALCdevice *device)
+static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err;
-
-    err = AudioOutputUnitStart(data->audioUnit);
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioOutputUnitStart failed\n");
@@ -358,64 +305,196 @@ static ALCboolean ca_start_playback(ALCdevice *device)
     return ALC_TRUE;
 }
 
-static void ca_stop_playback(ALCdevice *device)
+static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
+{
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+
+
+
+typedef struct ALCcoreAudioCapture {
+    DERIVE_FROM_TYPE(ALCbackend);
+
+    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;
+} ALCcoreAudioCapture;
+
+static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device);
+static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self);
+static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name);
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ALCboolean, reset)
+static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self);
+static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self);
+static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples);
+static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self);
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ClockLatency, getClockLatency)
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, lock)
+static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, unlock)
+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);
+    SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self);
+
+    self->audioUnit = 0;
+    self->audioConverter = NULL;
+    self->bufferList = NULL;
+    self->resampleBuffer = NULL;
+    self->ring = NULL;
+}
+
+static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self)
+{
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+
+    free(self->resampleBuffer);
+    self->resampleBuffer = NULL;
+
+    destroy_buffer_list(self->bufferList);
+    self->bufferList = NULL;
+
+    if(self->audioConverter)
+        AudioConverterDispose(self->audioConverter);
+    self->audioConverter = NULL;
+
+    if(self->audioUnit)
+        AudioComponentInstanceDispose(self->audioUnit);
+    self->audioUnit = 0;
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
+
+
+static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon,
+  AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
+  const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber),
+  UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData))
 {
-    ca_data *data = (ca_data*)device->ExtraData;
+    ALCcoreAudioCapture *self = inRefCon;
+    AudioUnitRenderActionFlags flags = 0;
     OSStatus err;
 
-    err = AudioOutputUnitStop(data->audioUnit);
+    // fill the bufferList with data from the input device
+    err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList);
     if(err != noErr)
-        ERR("AudioOutputUnitStop failed\n");
+    {
+        ERR("AudioUnitRender error: %d\n", err);
+        return err;
+    }
+
+    ll_ringbuffer_write(self->ring, self->bufferList->mBuffers[0].mData, inNumberFrames);
+
+    return noErr;
+}
+
+static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter),
+  UInt32 *ioNumberDataPackets, AudioBufferList *ioData,
+  AudioStreamPacketDescription** UNUSED(outDataPacketDescription),
+  void *inUserData)
+{
+    ALCcoreAudioCapture *self = inUserData;
+
+    // Read from the ring buffer and store temporarily in a large buffer
+    ll_ringbuffer_read(self->ring, self->resampleBuffer, *ioNumberDataPackets);
+
+    // Set the input data
+    ioData->mNumberBuffers = 1;
+    ioData->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
+    ioData->mBuffers[0].mData = self->resampleBuffer;
+    ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->format.mBytesPerFrame;
+
+    return noErr;
 }
 
-static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
+
+static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name)
 {
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
     AudioStreamBasicDescription requestedFormat;  // The application requested format
     AudioStreamBasicDescription hardwareFormat;   // The hardware format
     AudioStreamBasicDescription outputFormat;     // The AudioUnit output format
     AURenderCallbackStruct input;
-    ComponentDescription desc;
-    AudioDeviceID inputDevice;
+    AudioComponentDescription desc;
     UInt32 outputFrameCount;
     UInt32 propertySize;
+    AudioObjectPropertyAddress propertyAddress;
     UInt32 enableIO;
-    Component comp;
-    ca_data *data;
+    AudioComponent comp;
     OSStatus err;
 
-    if(!deviceName)
-        deviceName = ca_device;
-    else if(strcmp(deviceName, ca_device) != 0)
+    if(!name)
+        name = ca_device;
+    else if(strcmp(name, ca_device) != 0)
         return ALC_INVALID_VALUE;
 
     desc.componentType = kAudioUnitType_Output;
+#if TARGET_OS_IOS
+    desc.componentSubType = kAudioUnitSubType_RemoteIO;
+#else
     desc.componentSubType = kAudioUnitSubType_HALOutput;
+#endif
     desc.componentManufacturer = kAudioUnitManufacturer_Apple;
     desc.componentFlags = 0;
     desc.componentFlagsMask = 0;
 
     // Search for component with given description
-    comp = FindNextComponent(NULL, &desc);
+    comp = AudioComponentFindNext(NULL, &desc);
     if(comp == NULL)
     {
-        ERR("FindNextComponent failed\n");
+        ERR("AudioComponentFindNext failed\n");
         return ALC_INVALID_VALUE;
     }
 
-    data = calloc(1, sizeof(*data));
-    device->ExtraData = data;
-
     // Open the component
-    err = OpenAComponent(comp, &data->audioUnit);
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
     if(err != noErr)
     {
-        ERR("OpenAComponent failed\n");
+        ERR("AudioComponentInstanceNew failed\n");
         goto error;
     }
 
     // Turn off AudioUnit output
     enableIO = 0;
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -424,22 +503,28 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
 
     // Turn on AudioUnit input
     enableIO = 1;
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
         goto error;
     }
 
+#if !TARGET_OS_IOS
     // Get the default input device
+    AudioDeviceID inputDevice = kAudioDeviceUnknown;
+
     propertySize = sizeof(AudioDeviceID);
-    err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
+    propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+    propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
+    propertyAddress.mElement = kAudioObjectPropertyElementMaster;
+
+    err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
     if(err != noErr)
     {
-        ERR("AudioHardwareGetProperty failed\n");
+        ERR("AudioObjectGetPropertyData failed\n");
         goto error;
     }
-
     if(inputDevice == kAudioDeviceUnknown)
     {
         ERR("No input device found\n");
@@ -447,18 +532,19 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Track the input device
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
         goto error;
     }
+#endif
 
     // set capture callback
-    input.inputProc = ca_capture_callback;
-    input.inputProcRefCon = device;
+    input.inputProc = ALCcoreAudioCapture_RecordProc;
+    input.inputProcRefCon = self;
 
-    err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -466,7 +552,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Initialize the device
-    err = AudioUnitInitialize(data->audioUnit);
+    err = AudioUnitInitialize(self->audioUnit);
     if(err != noErr)
     {
         ERR("AudioUnitInitialize failed\n");
@@ -475,7 +561,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
 
     // Get the hardware format
     propertySize = sizeof(AudioStreamBasicDescription);
-    err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
     if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
     {
         ERR("AudioUnitGetProperty failed\n");
@@ -522,7 +608,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
         case DevFmtX51Rear:
         case DevFmtX61:
         case DevFmtX71:
-        case DevFmtBFormat3D:
+        case DevFmtAmbi3D:
             ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
             goto error;
     }
@@ -535,8 +621,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     requestedFormat.mFramesPerPacket = 1;
 
     // save requested format description for later use
-    data->format = requestedFormat;
-    data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
+    self->format = requestedFormat;
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
 
     // Use intermediate format for sample rate conversion (outputFormat)
     // Set sample rate to the same as hardware for resampling later
@@ -544,11 +630,11 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     outputFormat.mSampleRate = hardwareFormat.mSampleRate;
 
     // Determine sample rate ratio for resampling
-    data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
+    self->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
 
     // The output format should be the requested format, but using the hardware sample rate
     // This is because the AudioUnit will automatically scale other properties, except for sample rate
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed\n");
@@ -556,8 +642,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Set the AudioUnit output format frame count
-    outputFrameCount = device->UpdateSize * data->sampleRateRatio;
-    err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
+    outputFrameCount = device->UpdateSize * self->sampleRateRatio;
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
     if(err != noErr)
     {
         ERR("AudioUnitSetProperty failed: %d\n", err);
@@ -565,7 +651,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Set up sample converter
-    err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
+    err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter);
     if(err != noErr)
     {
         ERR("AudioConverterNew failed: %d\n", err);
@@ -573,92 +659,83 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
     }
 
     // Create a buffer for use in the resample callback
-    data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
+    self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
 
     // Allocate buffer for the AudioUnit output
-    data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
-    if(data->bufferList == NULL)
+    self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
+    if(self->bufferList == NULL)
         goto error;
 
-    data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
-    if(data->ring == NULL)
-        goto error;
+    self->ring = ll_ringbuffer_create(
+        (size_t)ceil(device->UpdateSize*self->sampleRateRatio*device->NumUpdates),
+        self->frameSize, false
+    );
+    if(!self->ring) goto error;
 
-    al_string_copy_cstr(&device->DeviceName, deviceName);
+    alstr_copy_cstr(&device->DeviceName, name);
 
     return ALC_NO_ERROR;
 
 error:
-    DestroyRingBuffer(data->ring);
-    free(data->resampleBuffer);
-    destroy_buffer_list(data->bufferList);
-
-    if(data->audioConverter)
-        AudioConverterDispose(data->audioConverter);
-    if(data->audioUnit)
-        CloseComponent(data->audioUnit);
-
-    free(data);
-    device->ExtraData = NULL;
+    ll_ringbuffer_free(self->ring);
+    self->ring = NULL;
+    free(self->resampleBuffer);
+    self->resampleBuffer = NULL;
+    destroy_buffer_list(self->bufferList);
+    self->bufferList = NULL;
+
+    if(self->audioConverter)
+        AudioConverterDispose(self->audioConverter);
+    self->audioConverter = NULL;
+    if(self->audioUnit)
+        AudioComponentInstanceDispose(self->audioUnit);
+    self->audioUnit = 0;
 
     return ALC_INVALID_VALUE;
 }
 
-static void ca_close_capture(ALCdevice *device)
-{
-    ca_data *data = (ca_data*)device->ExtraData;
-
-    DestroyRingBuffer(data->ring);
-    free(data->resampleBuffer);
-    destroy_buffer_list(data->bufferList);
 
-    AudioConverterDispose(data->audioConverter);
-    CloseComponent(data->audioUnit);
-
-    free(data);
-    device->ExtraData = NULL;
-}
-
-static void ca_start_capture(ALCdevice *device)
+static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err = AudioOutputUnitStart(data->audioUnit);
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
     if(err != noErr)
+    {
         ERR("AudioOutputUnitStart failed\n");
+        return ALC_FALSE;
+    }
+    return ALC_TRUE;
 }
 
-static void ca_stop_capture(ALCdevice *device)
+static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    OSStatus err = AudioOutputUnitStop(data->audioUnit);
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
     if(err != noErr)
         ERR("AudioOutputUnitStop failed\n");
 }
 
-static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
+static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
 {
-    ca_data *data = (ca_data*)device->ExtraData;
-    AudioBufferList *list;
+    union {
+        ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)];
+        AudioBufferList list;
+    } audiobuf = { { 0 } };
     UInt32 frameCount;
     OSStatus err;
 
     // If no samples are requested, just return
-    if(samples == 0)
-        return ALC_NO_ERROR;
-
-    // Allocate a temporary AudioBufferList to use as the return resamples data
-    list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
+    if(samples == 0) return ALC_NO_ERROR;
 
     // Point the resampling buffer to the capture buffer
-    list->mNumberBuffers = 1;
-    list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
-    list->mBuffers[0].mDataByteSize = samples * data->frameSize;
-    list->mBuffers[0].mData = buffer;
+    audiobuf.list.mNumberBuffers = 1;
+    audiobuf.list.mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
+    audiobuf.list.mBuffers[0].mDataByteSize = samples * self->frameSize;
+    audiobuf.list.mBuffers[0].mData = buffer;
 
     // Resample into another AudioBufferList
     frameCount = samples;
-    err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback,
-                                          device, &frameCount, list, NULL);
+    err = AudioConverterFillComplexBuffer(self->audioConverter,
+        ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL
+    );
     if(err != noErr)
     {
         ERR("AudioConverterFillComplexBuffer error: %d\n", err);
@@ -667,46 +744,73 @@ static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint sa
     return ALC_NO_ERROR;
 }
 
-static ALCuint ca_available_samples(ALCdevice *device)
+static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
 {
-    ca_data *data = device->ExtraData;
-    return RingBufferSize(data->ring) / data->sampleRateRatio;
+    return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
 }
 
 
-static const BackendFuncs ca_funcs = {
-    ca_open_playback,
-    ca_close_playback,
-    ca_reset_playback,
-    ca_start_playback,
-    ca_stop_playback,
-    ca_open_capture,
-    ca_close_capture,
-    ca_start_capture,
-    ca_stop_capture,
-    ca_capture_samples,
-    ca_available_samples
-};
-
-ALCboolean alc_ca_init(BackendFuncs *func_list)
+typedef struct ALCcoreAudioBackendFactory {
+    DERIVE_FROM_TYPE(ALCbackendFactory);
+} ALCcoreAudioBackendFactory;
+#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } }
+
+ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
+
+static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory *self);
+static DECLARE_FORWARD(ALCcoreAudioBackendFactory, ALCbackendFactory, void, deinit)
+static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory *self, ALCbackend_Type type);
+static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory *self, enum DevProbe type, al_string *outnames);
+static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
+DEFINE_ALCBACKENDFACTORY_VTABLE(ALCcoreAudioBackendFactory);
+
+
+ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void)
+{
+    static ALCcoreAudioBackendFactory factory = ALCCOREAUDIOBACKENDFACTORY_INITIALIZER;
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self))
 {
-    *func_list = ca_funcs;
     return ALC_TRUE;
 }
 
-void alc_ca_deinit(void)
+static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
 {
+    if(type == ALCbackend_Playback || ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
 }
 
-void alc_ca_probe(enum DevProbe type)
+static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
 {
     switch(type)
     {
         case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(ca_device);
-            break;
         case CAPTURE_DEVICE_PROBE:
-            AppendCaptureDeviceList(ca_device);
+            alstr_append_range(outnames, ca_device, ca_device+sizeof(ca_device));
             break;
     }
 }
+
+static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback)
+    {
+        ALCcoreAudioPlayback *backend;
+        NEW_OBJ(backend, ALCcoreAudioPlayback)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+    if(type == ALCbackend_Capture)
+    {
+        ALCcoreAudioCapture *backend;
+        NEW_OBJ(backend, ALCcoreAudioCapture)(device);
+        if(!backend) return NULL;
+        return STATIC_CAST(ALCbackend, backend);
+    }
+
+    return NULL;
+}