axmol/thirdparty/openal/alc/backends/coreaudio.cpp

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/**
* 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"
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#include "coreaudio.h"
#include <inttypes.h>
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#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
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#include <unistd.h>
#include <cmath>
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#include "alnumeric.h"
#include "core/converter.h"
#include "core/device.h"
#include "core/logging.h"
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#include "ringbuffer.h"
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
namespace {
static const char ca_device[] = "CoreAudio Default";
struct CoreAudioPlayback final : public BackendBase {
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CoreAudioPlayback(DeviceBase *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 char *name) override;
bool reset() override;
void start() override;
void stop() override;
AudioUnit mAudioUnit{};
uint 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
{
for(size_t i{0};i < ioData->mNumberBuffers;++i)
{
auto &buffer = ioData->mBuffers[i];
mDevice->renderSamples(buffer.mData, buffer.mDataByteSize/mFrameSize,
buffer.mNumberChannels);
}
return noErr;
}
void CoreAudioPlayback::open(const char *name)
{
if(!name)
name = ca_device;
else if(strcmp(name, ca_device) != 0)
throw al::backend_exception{al::backend_error::NoDevice, "Device name \"%s\" not found",
name};
/* open the default output unit */
AudioComponentDescription desc{};
desc.componentType = kAudioUnitType_Output;
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#if TARGET_OS_IOS || TARGET_OS_TV
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{al::backend_error::NoDevice, "Could not find audio component"};
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AudioUnit audioUnit{};
OSStatus err{AudioComponentInstanceNew(comp, &audioUnit)};
if(err != noErr)
throw al::backend_exception{al::backend_error::NoDevice,
"Could not create component instance: %u", err};
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err = AudioUnitInitialize(audioUnit);
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"Could not initialize audio unit: %u", err};
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/* WARNING: I don't know if "valid" audio unit values are guaranteed to be
* non-0. If not, this logic is broken.
*/
if(mAudioUnit)
{
AudioUnitUninitialize(mAudioUnit);
AudioComponentInstanceDispose(mAudioUnit);
}
mAudioUnit = audioUnit;
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<uint>(uint64_t{mDevice->BufferSize} *
streamFormat.mSampleRate / mDevice->Frequency);
mDevice->Frequency = static_cast<uint>(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();
/* 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;
}
void CoreAudioPlayback::start()
{
const OSStatus err{AudioOutputUnitStart(mAudioUnit)};
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"AudioOutputUnitStart failed: %d", err};
}
void CoreAudioPlayback::stop()
{
OSStatus err{AudioOutputUnitStop(mAudioUnit)};
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
struct CoreAudioCapture final : public BackendBase {
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CoreAudioCapture(DeviceBase *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 char *name) override;
void start() override;
void stop() override;
void captureSamples(al::byte *buffer, uint samples) override;
uint availableSamples() override;
AudioUnit mAudioUnit{0};
uint 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<uint>(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 char *name)
{
AudioStreamBasicDescription requestedFormat; // The application requested format
AudioStreamBasicDescription hardwareFormat; // The hardware format
AudioStreamBasicDescription outputFormat; // The AudioUnit output format
AURenderCallbackStruct input;
AudioComponentDescription desc;
UInt32 propertySize;
UInt32 enableIO;
AudioComponent comp;
OSStatus err;
if(!name)
name = ca_device;
else if(strcmp(name, ca_device) != 0)
throw al::backend_exception{al::backend_error::NoDevice, "Device name \"%s\" not found",
name};
desc.componentType = kAudioUnitType_Output;
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#if TARGET_OS_IOS || TARGET_OS_TV
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{al::backend_error::NoDevice, "Could not find audio component"};
// Open the component
err = AudioComponentInstanceNew(comp, &mAudioUnit);
if(err != noErr)
throw al::backend_exception{al::backend_error::NoDevice,
"Could not create component instance: %u", err};
// Turn off AudioUnit output
enableIO = 0;
err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output, 0, &enableIO, sizeof(enableIO));
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"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(enableIO));
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"Could not enable audio unit input property: %u", err};
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#if !TARGET_OS_IOS && !TARGET_OS_TV
{
// Get the default input device
AudioDeviceID inputDevice = kAudioDeviceUnknown;
propertySize = sizeof(AudioDeviceID);
AudioObjectPropertyAddress propertyAddress{};
propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
propertyAddress.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, nullptr,
&propertySize, &inputDevice);
if(err != noErr)
throw al::backend_exception{al::backend_error::NoDevice,
"Could not get input device: %u", err};
if(inputDevice == kAudioDeviceUnknown)
throw al::backend_exception{al::backend_error::NoDevice, "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{al::backend_error::NoDevice,
"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{al::backend_error::DeviceError,
"Could not set capture callback: %u", err};
// Disable buffer allocation for capture
UInt32 flag{0};
err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output, 1, &flag, sizeof(flag));
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"Could not disable buffer allocation property: %u", err};
// Initialize the device
err = AudioUnitInitialize(mAudioUnit);
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"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{al::backend_error::DeviceError,
"Could not get input format: %u", err};
// Set up the requested format description
switch(mDevice->FmtType)
{
case DevFmtByte:
requestedFormat.mBitsPerChannel = 8;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
break;
case DevFmtUByte:
requestedFormat.mBitsPerChannel = 8;
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
break;
case DevFmtShort:
requestedFormat.mBitsPerChannel = 16;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtUShort:
requestedFormat.mBitsPerChannel = 16;
requestedFormat.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtInt:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtUInt:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtFloat:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
}
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{al::backend_error::DeviceError, "%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{al::backend_error::DeviceError,
"Could not set input format: %u", err};
/* Calculate the minimum AudioUnit output format frame count for the pre-
* conversion ring buffer. Ensure at least 100ms for the total buffer.
*/
double srateScale{double{outputFormat.mSampleRate} / mDevice->Frequency};
auto FrameCount64 = maxu64(static_cast<uint64_t>(std::ceil(mDevice->BufferSize*srateScale)),
static_cast<UInt32>(outputFormat.mSampleRate)/10);
FrameCount64 += MaxResamplerPadding;
if(FrameCount64 > std::numeric_limits<int32_t>::max())
throw al::backend_exception{al::backend_error::DeviceError,
"Calculated frame count is too large: %" PRIu64, FrameCount64};
UInt32 outputFrameCount{};
propertySize = sizeof(outputFrameCount);
err = AudioUnitGetProperty(mAudioUnit, kAudioUnitProperty_MaximumFramesPerSlice,
kAudioUnitScope_Global, 0, &outputFrameCount, &propertySize);
if(err != noErr || propertySize != sizeof(outputFrameCount))
throw al::backend_exception{al::backend_error::DeviceError,
"Could not get input frame count: %u", err};
outputFrameCount = static_cast<UInt32>(maxu64(outputFrameCount, FrameCount64));
mRing = RingBuffer::Create(outputFrameCount, mFrameSize, false);
/* Set up sample converter if needed */
if(outputFormat.mSampleRate != mDevice->Frequency)
mConverter = CreateSampleConverter(mDevice->FmtType, mDevice->FmtType,
mFormat.mChannelsPerFrame, static_cast<uint>(hardwareFormat.mSampleRate),
mDevice->Frequency, Resampler::FastBSinc24);
mDevice->DeviceName = name;
}
void CoreAudioCapture::start()
{
OSStatus err{AudioOutputUnitStart(mAudioUnit)};
if(err != noErr)
throw al::backend_exception{al::backend_error::DeviceError,
"AudioOutputUnitStart failed: %d", err};
}
void CoreAudioCapture::stop()
{
OSStatus err{AudioOutputUnitStop(mAudioUnit)};
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
void CoreAudioCapture::captureSamples(al::byte *buffer, uint samples)
{
if(!mConverter)
{
mRing->read(buffer, samples);
return;
}
auto rec_vec = mRing->getReadVector();
const void *src0{rec_vec.first.buf};
auto src0len = static_cast<uint>(rec_vec.first.len);
uint 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<uint>(rec_vec.second.len);
got += mConverter->convert(&src1, &src1len, buffer + got*mFrameSize, samples-got);
total_read += rec_vec.second.len - src1len;
}
mRing->readAdvance(total_read);
}
uint CoreAudioCapture::availableSamples()
{
if(!mConverter) return static_cast<uint>(mRing->readSpace());
return mConverter->availableOut(static_cast<uint>(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; }
std::string CoreAudioBackendFactory::probe(BackendType type)
{
std::string outnames;
switch(type)
{
case BackendType::Playback:
case BackendType::Capture:
/* Includes null char. */
outnames.append(ca_device, sizeof(ca_device));
break;
}
return outnames;
}
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BackendPtr CoreAudioBackendFactory::createBackend(DeviceBase *device, BackendType type)
{
if(type == BackendType::Playback)
return BackendPtr{new CoreAudioPlayback{device}};
if(type == BackendType::Capture)
return BackendPtr{new CoreAudioCapture{device}};
return nullptr;
}