Convert the CoreAudio backend to C++

1 files changed, 820 insertions, 0 deletions

diff --git a/Alc/backends/coreaudio.cpp b/Alc/backends/coreaudio.cpp
new file mode 100644
index 00000000..08cdeeaa
--- /dev/null
+++ b/Alc/backends/coreaudio.cpp
@@ -0,0 +1,820 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "alMain.h"
+#include "alu.h"
+#include "ringbuffer.h"
+
+#include <unistd.h>
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+#include "backends/base.h"
+
+
+static const ALCchar ca_device[] = "CoreAudio Default";
+
+
+struct ALCcoreAudioPlayback final : public ALCbackend {
+    AudioUnit audioUnit;
+
+    ALuint frameSize;
+    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
+};
+
+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)
+
+DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback);
+
+
+static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device)
+{
+    new (self) ALCcoreAudioPlayback{};
+    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
+    SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self);
+
+    self->frameSize = 0;
+    memset(&self->format, 0, sizeof(self->format));
+}
+
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
+{
+    AudioUnitUninitialize(self->audioUnit);
+    AudioComponentInstanceDispose(self->audioUnit);
+
+    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+    self->~ALCcoreAudioPlayback();
+}
+
+
+static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
+  AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp),
+  UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData)
+{
+    ALCcoreAudioPlayback *self = static_cast<ALCcoreAudioPlayback*>(inRefCon);
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+
+    ALCcoreAudioPlayback_lock(self);
+    aluMixData(device, ioData->mBuffers[0].mData,
+               ioData->mBuffers[0].mDataByteSize / self->frameSize);
+    ALCcoreAudioPlayback_unlock(self);
+
+    return noErr;
+}
+
+
+static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioComponentDescription desc;
+    AudioComponent comp;
+    OSStatus err;
+
+    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 = AudioComponentFindNext(NULL, &desc);
+    if(comp == NULL)
+    {
+        ERR("AudioComponentFindNext failed\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioComponentInstanceNew failed\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    /* init and start the default audio unit... */
+    err = AudioUnitInitialize(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        AudioComponentInstanceDispose(self->audioUnit);
+        return ALC_INVALID_VALUE;
+    }
+
+    alstr_copy_cstr(&device->DeviceName, name);
+    return ALC_NO_ERROR;
+}
+
+static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
+{
+    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
+    AudioStreamBasicDescription streamFormat;
+    AURenderCallbackStruct input;
+    OSStatus err;
+    UInt32 size;
+
+    err = AudioUnitUninitialize(self->audioUnit);
+    if(err != noErr)
+        ERR("-- AudioUnitUninitialize failed.\n");
+
+    /* retrieve default output unit's properties (output side) */
+    size = sizeof(AudioStreamBasicDescription);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
+    if(err != noErr || size != sizeof(AudioStreamBasicDescription))
+    {
+        ERR("AudioUnitGetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+#if 0
+    TRACE("Output streamFormat of default output unit -\n");
+    TRACE("  streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
+    TRACE("  streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
+    TRACE("  streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
+    TRACE("  streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
+    TRACE("  streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
+    TRACE("  streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
+#endif
+
+    /* set default output unit's input side to match output side */
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+    if(device->Frequency != streamFormat.mSampleRate)
+    {
+        device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates *
+                                      streamFormat.mSampleRate /
+                                      device->Frequency);
+        device->Frequency = streamFormat.mSampleRate;
+    }
+
+    /* FIXME: How to tell what channels are what in the output device, and how
+     * to specify what we're giving?  eg, 6.0 vs 5.1 */
+    switch(streamFormat.mChannelsPerFrame)
+    {
+        case 1:
+            device->FmtChans = DevFmtMono;
+            break;
+        case 2:
+            device->FmtChans = DevFmtStereo;
+            break;
+        case 4:
+            device->FmtChans = DevFmtQuad;
+            break;
+        case 6:
+            device->FmtChans = DevFmtX51;
+            break;
+        case 7:
+            device->FmtChans = DevFmtX61;
+            break;
+        case 8:
+            device->FmtChans = DevFmtX71;
+            break;
+        default:
+            ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
+            device->FmtChans = DevFmtStereo;
+            streamFormat.mChannelsPerFrame = 2;
+            break;
+    }
+    SetDefaultWFXChannelOrder(device);
+
+    /* use channel count and sample rate from the default output unit's current
+     * parameters, but reset everything else */
+    streamFormat.mFramesPerPacket = 1;
+    streamFormat.mFormatFlags = 0;
+    switch(device->FmtType)
+    {
+        case DevFmtUByte:
+            device->FmtType = DevFmtByte;
+            /* fall-through */
+        case DevFmtByte:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 8;
+            break;
+        case DevFmtUShort:
+            device->FmtType = DevFmtShort;
+            /* fall-through */
+        case DevFmtShort:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 16;
+            break;
+        case DevFmtUInt:
+            device->FmtType = DevFmtInt;
+            /* fall-through */
+        case DevFmtInt:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 32;
+            break;
+        case DevFmtFloat:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
+            streamFormat.mBitsPerChannel = 32;
+            break;
+    }
+    streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame *
+                                  streamFormat.mBitsPerChannel / 8;
+    streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame;
+    streamFormat.mFormatID = kAudioFormatLinearPCM;
+    streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
+                                 kLinearPCMFormatFlagIsPacked;
+
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+    /* setup callback */
+    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
+    input.inputProc = ALCcoreAudioPlayback_MixerProc;
+    input.inputProcRefCon = self;
+
+    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return ALC_FALSE;
+    }
+
+    /* init the default audio unit... */
+    err = AudioUnitInitialize(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
+{
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioOutputUnitStart failed\n");
+        return ALC_FALSE;
+    }
+
+    return ALC_TRUE;
+}
+
+static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
+{
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+
+struct ALCcoreAudioCapture final : public 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;
+};
+
+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 = static_cast<AudioBufferList*>(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)
+{
+    new (self) ALCcoreAudioCapture{};
+    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));
+    self->~ALCcoreAudioCapture();
+}
+
+
+static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon,
+  AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
+  const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber),
+  UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData))
+{
+    ALCcoreAudioCapture *self = static_cast<ALCcoreAudioCapture*>(inRefCon);
+    AudioUnitRenderActionFlags flags = 0;
+    OSStatus err;
+
+    // fill the bufferList with data from the input device
+    err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList);
+    if(err != noErr)
+    {
+        ERR("AudioUnitRender error: %d\n", err);
+        return err;
+    }
+
+    ll_ringbuffer_write(self->ring, static_cast<const char*>(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 = reinterpret_cast<ALCcoreAudioCapture*>(inUserData);
+
+    // Read from the ring buffer and store temporarily in a large buffer
+    ll_ringbuffer_read(self->ring, static_cast<char*>(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 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;
+    AudioComponentDescription desc;
+    UInt32 outputFrameCount;
+    UInt32 propertySize;
+    AudioObjectPropertyAddress propertyAddress;
+    UInt32 enableIO;
+    AudioComponent comp;
+    OSStatus err;
+
+    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 = AudioComponentFindNext(NULL, &desc);
+    if(comp == NULL)
+    {
+        ERR("AudioComponentFindNext failed\n");
+        return ALC_INVALID_VALUE;
+    }
+
+    // Open the component
+    err = AudioComponentInstanceNew(comp, &self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioComponentInstanceNew failed\n");
+        goto error;
+    }
+
+    // Turn off AudioUnit output
+    enableIO = 0;
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Turn on AudioUnit input
+    enableIO = 1;
+    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);
+        propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+        propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
+        propertyAddress.mElement = kAudioObjectPropertyElementMaster;
+
+        err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
+        if(err != noErr)
+        {
+            ERR("AudioObjectGetPropertyData failed\n");
+            goto error;
+        }
+        if(inputDevice == kAudioDeviceUnknown)
+        {
+            ERR("No input device found\n");
+            goto error;
+        }
+
+        // Track the input device
+        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 = ALCcoreAudioCapture_RecordProc;
+    input.inputProcRefCon = self;
+
+    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Initialize the device
+    err = AudioUnitInitialize(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        goto error;
+    }
+
+    // Get the hardware format
+    propertySize = sizeof(AudioStreamBasicDescription);
+    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
+    if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
+    {
+        ERR("AudioUnitGetProperty failed\n");
+        goto error;
+    }
+
+    // Set up the requested format description
+    switch(device->FmtType)
+    {
+        case DevFmtUByte:
+            requestedFormat.mBitsPerChannel = 8;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtShort:
+            requestedFormat.mBitsPerChannel = 16;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtInt:
+            requestedFormat.mBitsPerChannel = 32;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtFloat:
+            requestedFormat.mBitsPerChannel = 32;
+            requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
+            break;
+        case DevFmtByte:
+        case DevFmtUShort:
+        case DevFmtUInt:
+            ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
+            goto error;
+    }
+
+    switch(device->FmtChans)
+    {
+        case DevFmtMono:
+            requestedFormat.mChannelsPerFrame = 1;
+            break;
+        case DevFmtStereo:
+            requestedFormat.mChannelsPerFrame = 2;
+            break;
+
+        case DevFmtQuad:
+        case DevFmtX51:
+        case DevFmtX51Rear:
+        case DevFmtX61:
+        case DevFmtX71:
+        case DevFmtAmbi3D:
+            ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
+            goto error;
+    }
+
+    requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
+    requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
+    requestedFormat.mSampleRate = device->Frequency;
+    requestedFormat.mFormatID = kAudioFormatLinearPCM;
+    requestedFormat.mReserved = 0;
+    requestedFormat.mFramesPerPacket = 1;
+
+    // save requested format description for later use
+    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
+    outputFormat = requestedFormat;
+    outputFormat.mSampleRate = hardwareFormat.mSampleRate;
+
+    // Determine sample rate ratio for resampling
+    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(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        goto error;
+    }
+
+    // Set the AudioUnit output format frame count
+    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);
+        goto error;
+    }
+
+    // Set up sample converter
+    err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter);
+    if(err != noErr)
+    {
+        ERR("AudioConverterNew failed: %d\n", err);
+        goto error;
+    }
+
+    // Create a buffer for use in the resample callback
+    self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
+
+    // Allocate buffer for the AudioUnit output
+    self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
+    if(self->bufferList == 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;
+
+    alstr_copy_cstr(&device->DeviceName, name);
+
+    return ALC_NO_ERROR;
+
+error:
+    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 ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
+{
+    OSStatus err = AudioOutputUnitStart(self->audioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioOutputUnitStart failed\n");
+        return ALC_FALSE;
+    }
+    return ALC_TRUE;
+}
+
+static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
+{
+    OSStatus err = AudioOutputUnitStop(self->audioUnit);
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
+{
+    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;
+
+    // Point the resampling buffer to the capture 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(self->audioConverter,
+        ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL
+    );
+    if(err != noErr)
+    {
+        ERR("AudioConverterFillComplexBuffer error: %d\n", err);
+        return ALC_INVALID_VALUE;
+    }
+    return ALC_NO_ERROR;
+}
+
+static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
+{
+    return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
+}
+
+
+struct ALCcoreAudioBackendFactory final : public ALCbackendFactory {
+    ALCcoreAudioBackendFactory() noexcept;
+};
+
+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);
+
+
+ALCcoreAudioBackendFactory::ALCcoreAudioBackendFactory() noexcept
+  : ALCbackendFactory{GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory)}
+{ }
+
+ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void)
+{
+    static ALCcoreAudioBackendFactory factory{};
+    return STATIC_CAST(ALCbackendFactory, &factory);
+}
+
+
+static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self))
+{
+    return ALC_TRUE;
+}
+
+static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
+{
+    if(type == ALCbackend_Playback || ALCbackend_Capture)
+        return ALC_TRUE;
+    return ALC_FALSE;
+}
+
+static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
+{
+    switch(type)
+    {
+        case ALL_DEVICE_PROBE:
+        case CAPTURE_DEVICE_PROBE:
+            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;
+}