1 files changed, 659 insertions, 0 deletions
diff --git a/alc/backends/coreaudio.cpp b/alc/backends/coreaudio.cpp
new file mode 100644
index 00000000..7c18287b
--- /dev/null
+++ b/alc/backends/coreaudio.cpp
@@ -0,0 +1,659 @@
+/**
+ * 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 "backends/coreaudio.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "alcmain.h"
+#include "alexcpt.h"
+#include "alu.h"
+#include "ringbuffer.h"
+#include "converter.h"
+#include "backends/base.h"
+
+#include <unistd.h>
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+
+namespace {
+
+static const ALCchar ca_device[] = "CoreAudio Default";
+
+
+struct CoreAudioPlayback final : public BackendBase {
+    CoreAudioPlayback(ALCdevice *device) noexcept : BackendBase{device} { }
+    ~CoreAudioPlayback() override;
+
+    OSStatus MixerProc(AudioUnitRenderActionFlags *ioActionFlags,
+        const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames,
+        AudioBufferList *ioData) noexcept;
+    static OSStatus MixerProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
+        const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames,
+        AudioBufferList *ioData) noexcept
+    {
+        return static_cast<CoreAudioPlayback*>(inRefCon)->MixerProc(ioActionFlags, inTimeStamp,
+            inBusNumber, inNumberFrames, ioData);
+    }
+
+    void open(const ALCchar *name) override;
+    bool reset() override;
+    bool start() override;
+    void stop() override;
+
+    AudioUnit mAudioUnit{};
+
+    ALuint mFrameSize{0u};
+    AudioStreamBasicDescription mFormat{}; // This is the OpenAL format as a CoreAudio ASBD
+
+    DEF_NEWDEL(CoreAudioPlayback)
+};
+
+CoreAudioPlayback::~CoreAudioPlayback()
+{
+    AudioUnitUninitialize(mAudioUnit);
+    AudioComponentInstanceDispose(mAudioUnit);
+}
+
+
+OSStatus CoreAudioPlayback::MixerProc(AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32,
+    UInt32, AudioBufferList *ioData) noexcept
+{
+    std::lock_guard<CoreAudioPlayback> _{*this};
+    aluMixData(mDevice, ioData->mBuffers[0].mData, ioData->mBuffers[0].mDataByteSize/mFrameSize);
+    return noErr;
+}
+
+
+void CoreAudioPlayback::open(const ALCchar *name)
+{
+    if(!name)
+        name = ca_device;
+    else if(strcmp(name, ca_device) != 0)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Device name \"%s\" not found", name};
+
+    /* open the default output unit */
+    AudioComponentDescription desc{};
+    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;
+
+    AudioComponent comp{AudioComponentFindNext(NULL, &desc)};
+    if(comp == nullptr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not find audio component"};
+
+    OSStatus err{AudioComponentInstanceNew(comp, &mAudioUnit)};
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not create component instance: %u",
+            err};
+
+    /* init and start the default audio unit... */
+    err = AudioUnitInitialize(mAudioUnit);
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not initialize audio unit: %u", err};
+
+    mDevice->DeviceName = name;
+}
+
+bool CoreAudioPlayback::reset()
+{
+    OSStatus err{AudioUnitUninitialize(mAudioUnit)};
+    if(err != noErr)
+        ERR("-- AudioUnitUninitialize failed.\n");
+
+    /* retrieve default output unit's properties (output side) */
+    AudioStreamBasicDescription streamFormat{};
+    auto size = static_cast<UInt32>(sizeof(AudioStreamBasicDescription));
+    err = AudioUnitGetProperty(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
+        0, &streamFormat, &size);
+    if(err != noErr || size != sizeof(AudioStreamBasicDescription))
+    {
+        ERR("AudioUnitGetProperty failed\n");
+        return 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(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
+        0, &streamFormat, size);
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return false;
+    }
+
+    if(mDevice->Frequency != streamFormat.mSampleRate)
+    {
+        mDevice->BufferSize = static_cast<ALuint>(uint64_t{mDevice->BufferSize} *
+            streamFormat.mSampleRate / mDevice->Frequency);
+        mDevice->Frequency = static_cast<ALuint>(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:
+            mDevice->FmtChans = DevFmtMono;
+            break;
+        case 2:
+            mDevice->FmtChans = DevFmtStereo;
+            break;
+        case 4:
+            mDevice->FmtChans = DevFmtQuad;
+            break;
+        case 6:
+            mDevice->FmtChans = DevFmtX51;
+            break;
+        case 7:
+            mDevice->FmtChans = DevFmtX61;
+            break;
+        case 8:
+            mDevice->FmtChans = DevFmtX71;
+            break;
+        default:
+            ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
+            mDevice->FmtChans = DevFmtStereo;
+            streamFormat.mChannelsPerFrame = 2;
+            break;
+    }
+    SetDefaultWFXChannelOrder(mDevice);
+
+    /* 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(mDevice->FmtType)
+    {
+        case DevFmtUByte:
+            mDevice->FmtType = DevFmtByte;
+            /* fall-through */
+        case DevFmtByte:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 8;
+            break;
+        case DevFmtUShort:
+            mDevice->FmtType = DevFmtShort;
+            /* fall-through */
+        case DevFmtShort:
+            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
+            streamFormat.mBitsPerChannel = 16;
+            break;
+        case DevFmtUInt:
+            mDevice->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(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
+        0, &streamFormat, sizeof(AudioStreamBasicDescription));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return false;
+    }
+
+    /* setup callback */
+    mFrameSize = mDevice->frameSizeFromFmt();
+    AURenderCallbackStruct input{};
+    input.inputProc = CoreAudioPlayback::MixerProcC;
+    input.inputProcRefCon = this;
+
+    err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_SetRenderCallback,
+        kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
+    if(err != noErr)
+    {
+        ERR("AudioUnitSetProperty failed\n");
+        return false;
+    }
+
+    /* init the default audio unit... */
+    err = AudioUnitInitialize(mAudioUnit);
+    if(err != noErr)
+    {
+        ERR("AudioUnitInitialize failed\n");
+        return false;
+    }
+
+    return true;
+}
+
+bool CoreAudioPlayback::start()
+{
+    OSStatus err{AudioOutputUnitStart(mAudioUnit)};
+    if(err != noErr)
+    {
+        ERR("AudioOutputUnitStart failed\n");
+        return false;
+    }
+    return true;
+}
+
+void CoreAudioPlayback::stop()
+{
+    OSStatus err{AudioOutputUnitStop(mAudioUnit)};
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+
+struct CoreAudioCapture final : public BackendBase {
+    CoreAudioCapture(ALCdevice *device) noexcept : BackendBase{device} { }
+    ~CoreAudioCapture() override;
+
+    OSStatus RecordProc(AudioUnitRenderActionFlags *ioActionFlags,
+        const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
+        UInt32 inNumberFrames, AudioBufferList *ioData) noexcept;
+    static OSStatus RecordProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
+        const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames,
+        AudioBufferList *ioData) noexcept
+    {
+        return static_cast<CoreAudioCapture*>(inRefCon)->RecordProc(ioActionFlags, inTimeStamp,
+            inBusNumber, inNumberFrames, ioData);
+    }
+
+    void open(const ALCchar *name) override;
+    bool start() override;
+    void stop() override;
+    ALCenum captureSamples(al::byte *buffer, ALCuint samples) override;
+    ALCuint availableSamples() override;
+
+    AudioUnit mAudioUnit{0};
+
+    ALuint mFrameSize{0u};
+    AudioStreamBasicDescription mFormat{};  // This is the OpenAL format as a CoreAudio ASBD
+
+    SampleConverterPtr mConverter;
+
+    RingBufferPtr mRing{nullptr};
+
+    DEF_NEWDEL(CoreAudioCapture)
+};
+
+CoreAudioCapture::~CoreAudioCapture()
+{
+    if(mAudioUnit)
+        AudioComponentInstanceDispose(mAudioUnit);
+    mAudioUnit = 0;
+}
+
+
+OSStatus CoreAudioCapture::RecordProc(AudioUnitRenderActionFlags*,
+    const AudioTimeStamp *inTimeStamp, UInt32, UInt32 inNumberFrames,
+    AudioBufferList*) noexcept
+{
+    AudioUnitRenderActionFlags flags = 0;
+    union {
+        al::byte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)*2];
+        AudioBufferList list;
+    } audiobuf{};
+
+    auto rec_vec = mRing->getWriteVector();
+    inNumberFrames = static_cast<UInt32>(minz(inNumberFrames,
+        rec_vec.first.len+rec_vec.second.len));
+
+    // Fill the ringbuffer's two segments with data from the input device
+    if(rec_vec.first.len >= inNumberFrames)
+    {
+        audiobuf.list.mNumberBuffers = 1;
+        audiobuf.list.mBuffers[0].mNumberChannels = mFormat.mChannelsPerFrame;
+        audiobuf.list.mBuffers[0].mData = rec_vec.first.buf;
+        audiobuf.list.mBuffers[0].mDataByteSize = inNumberFrames * mFormat.mBytesPerFrame;
+    }
+    else
+    {
+        const auto remaining = static_cast<ALuint>(inNumberFrames - rec_vec.first.len);
+        audiobuf.list.mNumberBuffers = 2;
+        audiobuf.list.mBuffers[0].mNumberChannels = mFormat.mChannelsPerFrame;
+        audiobuf.list.mBuffers[0].mData = rec_vec.first.buf;
+        audiobuf.list.mBuffers[0].mDataByteSize = static_cast<UInt32>(rec_vec.first.len) *
+            mFormat.mBytesPerFrame;
+        audiobuf.list.mBuffers[1].mNumberChannels = mFormat.mChannelsPerFrame;
+        audiobuf.list.mBuffers[1].mData = rec_vec.second.buf;
+        audiobuf.list.mBuffers[1].mDataByteSize = remaining * mFormat.mBytesPerFrame;
+    }
+    OSStatus err{AudioUnitRender(mAudioUnit, &flags, inTimeStamp, audiobuf.list.mNumberBuffers,
+        inNumberFrames, &audiobuf.list)};
+    if(err != noErr)
+    {
+        ERR("AudioUnitRender error: %d\n", err);
+        return err;
+    }
+
+    mRing->writeAdvance(inNumberFrames);
+    return noErr;
+}
+
+
+void CoreAudioCapture::open(const ALCchar *name)
+{
+    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;
+#if !TARGET_OS_IOS
+    AudioObjectPropertyAddress propertyAddress;
+#endif
+    UInt32 enableIO;
+    AudioComponent comp;
+    OSStatus err;
+
+    if(!name)
+        name = ca_device;
+    else if(strcmp(name, ca_device) != 0)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Device name \"%s\" not found", name};
+
+    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)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not find audio component"};
+
+    // Open the component
+    err = AudioComponentInstanceNew(comp, &mAudioUnit);
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not create component instance: %u",
+            err};
+
+    // Turn off AudioUnit output
+    enableIO = 0;
+    err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO,
+        kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE,
+            "Could not disable audio unit output property: %u", err};
+
+    // Turn on AudioUnit input
+    enableIO = 1;
+    err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO,
+        kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE,
+            "Could not enable audio unit input property: %u", err};
+
+#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, nullptr,
+            &propertySize, &inputDevice);
+        if(err != noErr)
+            throw al::backend_exception{ALC_INVALID_VALUE, "Could not get input device: %u", err};
+        if(inputDevice == kAudioDeviceUnknown)
+            throw al::backend_exception{ALC_INVALID_VALUE, "Unknown input device"};
+
+        // Track the input device
+        err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_CurrentDevice,
+            kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
+        if(err != noErr)
+            throw al::backend_exception{ALC_INVALID_VALUE, "Could not set input device: %u", err};
+    }
+#endif
+
+    // set capture callback
+    input.inputProc = CoreAudioCapture::RecordProcC;
+    input.inputProcRefCon = this;
+
+    err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_SetInputCallback,
+        kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not set capture callback: %u", err};
+
+    // Initialize the device
+    err = AudioUnitInitialize(mAudioUnit);
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not initialize audio unit: %u", err};
+
+    // Get the hardware format
+    propertySize = sizeof(AudioStreamBasicDescription);
+    err = AudioUnitGetProperty(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
+        1, &hardwareFormat, &propertySize);
+    if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not get input format: %u", err};
+
+    // Set up the requested format description
+    switch(mDevice->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:
+        throw al::backend_exception{ALC_INVALID_VALUE, "%s samples not suppoted",
+            DevFmtTypeString(mDevice->FmtType)};
+    }
+
+    switch(mDevice->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:
+        throw al::backend_exception{ALC_INVALID_VALUE, "%s not supported",
+            DevFmtChannelsString(mDevice->FmtChans)};
+    }
+
+    requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
+    requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
+    requestedFormat.mSampleRate = mDevice->Frequency;
+    requestedFormat.mFormatID = kAudioFormatLinearPCM;
+    requestedFormat.mReserved = 0;
+    requestedFormat.mFramesPerPacket = 1;
+
+    // save requested format description for later use
+    mFormat = requestedFormat;
+    mFrameSize = mDevice->frameSizeFromFmt();
+
+    // 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;
+
+    // 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(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
+        1, &outputFormat, sizeof(outputFormat));
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Could not set input format: %u", err};
+
+    // Set the AudioUnit output format frame count
+    uint64_t FrameCount64{mDevice->UpdateSize};
+    FrameCount64 = static_cast<uint64_t>(FrameCount64*outputFormat.mSampleRate + mDevice->Frequency-1) /
+        mDevice->Frequency;
+    FrameCount64 += MAX_RESAMPLER_PADDING;
+    if(FrameCount64 > std::numeric_limits<uint32_t>::max()/2)
+        throw al::backend_exception{ALC_INVALID_VALUE,
+            "Calculated frame count is too large: %" PRIu64, FrameCount64};
+
+    outputFrameCount = static_cast<uint32_t>(FrameCount64);
+    err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_MaximumFramesPerSlice,
+        kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
+    if(err != noErr)
+        throw al::backend_exception{ALC_INVALID_VALUE, "Failed to set capture frame count: %u",
+            err};
+
+    // Set up sample converter if needed
+    if(outputFormat.mSampleRate != mDevice->Frequency)
+        mConverter = CreateSampleConverter(mDevice->FmtType, mDevice->FmtType,
+            mFormat.mChannelsPerFrame, static_cast<ALuint>(hardwareFormat.mSampleRate),
+            mDevice->Frequency, Resampler::FastBSinc24);
+
+    mRing = CreateRingBuffer(outputFrameCount, mFrameSize, false);
+
+    mDevice->DeviceName = name;
+}
+
+
+bool CoreAudioCapture::start()
+{
+    OSStatus err{AudioOutputUnitStart(mAudioUnit)};
+    if(err != noErr)
+    {
+        ERR("AudioOutputUnitStart failed\n");
+        return false;
+    }
+    return true;
+}
+
+void CoreAudioCapture::stop()
+{
+    OSStatus err{AudioOutputUnitStop(mAudioUnit)};
+    if(err != noErr)
+        ERR("AudioOutputUnitStop failed\n");
+}
+
+ALCenum CoreAudioCapture::captureSamples(al::byte *buffer, ALCuint samples)
+{
+    if(!mConverter)
+    {
+        mRing->read(buffer, samples);
+        return ALC_NO_ERROR;
+    }
+
+    auto rec_vec = mRing->getReadVector();
+    const void *src0{rec_vec.first.buf};
+    auto src0len = static_cast<ALuint>(rec_vec.first.len);
+    ALuint got{mConverter->convert(&src0, &src0len, buffer, samples)};
+    size_t total_read{rec_vec.first.len - src0len};
+    if(got < samples && !src0len && rec_vec.second.len > 0)
+    {
+        const void *src1{rec_vec.second.buf};
+        auto src1len = static_cast<ALuint>(rec_vec.second.len);
+        got += mConverter->convert(&src1, &src1len, buffer+got, samples-got);
+        total_read += rec_vec.second.len - src1len;
+    }
+
+    mRing->readAdvance(total_read);
+    return ALC_NO_ERROR;
+}
+
+ALCuint CoreAudioCapture::availableSamples()
+{
+    if(!mConverter) return static_cast<ALCuint>(mRing->readSpace());
+    return mConverter->availableOut(static_cast<ALCuint>(mRing->readSpace()));
+}
+
+} // namespace
+
+BackendFactory &CoreAudioBackendFactory::getFactory()
+{
+    static CoreAudioBackendFactory factory{};
+    return factory;
+}
+
+bool CoreAudioBackendFactory::init() { return true; }
+
+bool CoreAudioBackendFactory::querySupport(BackendType type)
+{ return type == BackendType::Playback || type == BackendType::Capture; }
+
+void CoreAudioBackendFactory::probe(DevProbe type, std::string *outnames)
+{
+    switch(type)
+    {
+        case DevProbe::Playback:
+        case DevProbe::Capture:
+            /* Includes null char. */
+            outnames->append(ca_device, sizeof(ca_device));
+            break;
+    }
+}
+
+BackendPtr CoreAudioBackendFactory::createBackend(ALCdevice *device, BackendType type)
+{
+    if(type == BackendType::Playback)
+        return BackendPtr{new CoreAudioPlayback{device}};
+    if(type == BackendType::Capture)
+        return BackendPtr{new CoreAudioCapture{device}};
+    return nullptr;
+}