/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include "backends/coreaudio.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "alMain.h"
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#include "alu.h"
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#include "ringbuffer.h"
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#include "converter.h"
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#include "backends/base.h"
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#include <unistd.h>
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#include <AudioUnit/AudioUnit.h>
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#include <AudioToolbox/AudioToolbox.h>
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namespace {
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static const ALCchar ca_device[] = "CoreAudio Default";
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struct CoreAudioPlayback final : public BackendBase {
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CoreAudioPlayback(ALCdevice *device) noexcept : BackendBase{device} { }
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~CoreAudioPlayback() override;
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static OSStatus MixerProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames,
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AudioBufferList *ioData);
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OSStatus MixerProc(AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames,
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AudioBufferList *ioData);
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ALCenum open(const ALCchar *name) override;
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ALCboolean reset() override;
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ALCboolean start() override;
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void stop() override;
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AudioUnit mAudioUnit;
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ALuint mFrameSize{0u};
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AudioStreamBasicDescription mFormat{}; // This is the OpenAL format as a CoreAudio ASBD
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static constexpr inline const char *CurrentPrefix() noexcept { return "CoreAudioPlayback::"; }
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DEF_NEWDEL(CoreAudioPlayback)
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};
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CoreAudioPlayback::~CoreAudioPlayback()
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{
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AudioUnitUninitialize(mAudioUnit);
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AudioComponentInstanceDispose(mAudioUnit);
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}
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OSStatus CoreAudioPlayback::MixerProcC(void *inRefCon,
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AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
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{
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return static_cast<CoreAudioPlayback*>(inRefCon)->MixerProc(ioActionFlags, inTimeStamp,
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inBusNumber, inNumberFrames, ioData);
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}
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OSStatus CoreAudioPlayback::MixerProc(AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
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const AudioTimeStamp* UNUSED(inTimeStamp), UInt32 UNUSED(inBusNumber),
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UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData)
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{
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lock();
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aluMixData(mDevice, ioData->mBuffers[0].mData, ioData->mBuffers[0].mDataByteSize/mFrameSize);
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unlock();
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return noErr;
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}
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ALCenum CoreAudioPlayback::open(const ALCchar *name)
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{
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if(!name)
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name = ca_device;
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else if(strcmp(name, ca_device) != 0)
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return ALC_INVALID_VALUE;
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/* open the default output unit */
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AudioComponentDescription desc{};
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desc.componentType = kAudioUnitType_Output;
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#if TARGET_OS_IOS
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desc.componentSubType = kAudioUnitSubType_RemoteIO;
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#else
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desc.componentSubType = kAudioUnitSubType_DefaultOutput;
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#endif
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desc.componentManufacturer = kAudioUnitManufacturer_Apple;
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desc.componentFlags = 0;
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desc.componentFlagsMask = 0;
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AudioComponent comp{AudioComponentFindNext(NULL, &desc)};
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if(comp == nullptr)
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{
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ERR("AudioComponentFindNext failed\n");
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return ALC_INVALID_VALUE;
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}
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OSStatus err{AudioComponentInstanceNew(comp, &mAudioUnit)};
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if(err != noErr)
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{
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ERR("AudioComponentInstanceNew failed\n");
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return ALC_INVALID_VALUE;
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}
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/* init and start the default audio unit... */
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err = AudioUnitInitialize(mAudioUnit);
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if(err != noErr)
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{
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ERR("AudioUnitInitialize failed\n");
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AudioComponentInstanceDispose(mAudioUnit);
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return ALC_INVALID_VALUE;
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}
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mDevice->DeviceName = name;
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return ALC_NO_ERROR;
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}
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ALCboolean CoreAudioPlayback::reset()
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{
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OSStatus err{AudioUnitUninitialize(mAudioUnit)};
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if(err != noErr)
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ERR("-- AudioUnitUninitialize failed.\n");
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/* retrieve default output unit's properties (output side) */
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AudioStreamBasicDescription streamFormat{};
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auto size = static_cast<UInt32>(sizeof(AudioStreamBasicDescription));
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err = AudioUnitGetProperty(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
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0, &streamFormat, &size);
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if(err != noErr || size != sizeof(AudioStreamBasicDescription))
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{
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ERR("AudioUnitGetProperty failed\n");
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return ALC_FALSE;
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}
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#if 0
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TRACE("Output streamFormat of default output unit -\n");
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TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
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TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
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TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
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TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
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TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
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TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
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#endif
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/* set default output unit's input side to match output side */
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err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
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0, &streamFormat, size);
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_FALSE;
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}
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if(mDevice->Frequency != streamFormat.mSampleRate)
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{
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mDevice->BufferSize = static_cast<ALuint>(uint64_t{mDevice->BufferSize} *
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streamFormat.mSampleRate / mDevice->Frequency);
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mDevice->Frequency = streamFormat.mSampleRate;
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}
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/* FIXME: How to tell what channels are what in the output device, and how
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* to specify what we're giving? eg, 6.0 vs 5.1 */
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switch(streamFormat.mChannelsPerFrame)
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{
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case 1:
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mDevice->FmtChans = DevFmtMono;
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break;
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case 2:
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mDevice->FmtChans = DevFmtStereo;
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break;
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case 4:
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mDevice->FmtChans = DevFmtQuad;
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break;
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case 6:
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mDevice->FmtChans = DevFmtX51;
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break;
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case 7:
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mDevice->FmtChans = DevFmtX61;
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break;
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case 8:
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mDevice->FmtChans = DevFmtX71;
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break;
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default:
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ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
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mDevice->FmtChans = DevFmtStereo;
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streamFormat.mChannelsPerFrame = 2;
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break;
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}
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SetDefaultWFXChannelOrder(mDevice);
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/* use channel count and sample rate from the default output unit's current
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* parameters, but reset everything else */
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streamFormat.mFramesPerPacket = 1;
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streamFormat.mFormatFlags = 0;
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switch(mDevice->FmtType)
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{
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case DevFmtUByte:
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mDevice->FmtType = DevFmtByte;
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/* fall-through */
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case DevFmtByte:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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streamFormat.mBitsPerChannel = 8;
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break;
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case DevFmtUShort:
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mDevice->FmtType = DevFmtShort;
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/* fall-through */
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case DevFmtShort:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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streamFormat.mBitsPerChannel = 16;
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break;
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case DevFmtUInt:
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mDevice->FmtType = DevFmtInt;
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/* fall-through */
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case DevFmtInt:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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streamFormat.mBitsPerChannel = 32;
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break;
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case DevFmtFloat:
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streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
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streamFormat.mBitsPerChannel = 32;
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break;
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}
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streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame *
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streamFormat.mBitsPerChannel / 8;
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streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame;
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streamFormat.mFormatID = kAudioFormatLinearPCM;
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streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
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kLinearPCMFormatFlagIsPacked;
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err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
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0, &streamFormat, sizeof(AudioStreamBasicDescription));
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_FALSE;
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}
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/* setup callback */
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mFrameSize = mDevice->frameSizeFromFmt();
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AURenderCallbackStruct input{};
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input.inputProc = CoreAudioPlayback::MixerProcC;
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input.inputProcRefCon = this;
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err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_SetRenderCallback,
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kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_FALSE;
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}
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/* init the default audio unit... */
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err = AudioUnitInitialize(mAudioUnit);
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if(err != noErr)
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{
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ERR("AudioUnitInitialize failed\n");
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return ALC_FALSE;
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}
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return ALC_TRUE;
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}
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ALCboolean CoreAudioPlayback::start()
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{
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OSStatus err{AudioOutputUnitStart(mAudioUnit)};
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if(err != noErr)
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{
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ERR("AudioOutputUnitStart failed\n");
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return ALC_FALSE;
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}
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return ALC_TRUE;
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}
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void CoreAudioPlayback::stop()
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{
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OSStatus err{AudioOutputUnitStop(mAudioUnit)};
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if(err != noErr)
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ERR("AudioOutputUnitStop failed\n");
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}
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struct CoreAudioCapture final : public BackendBase {
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CoreAudioCapture(ALCdevice *device) noexcept : BackendBase{device} { }
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~CoreAudioCapture() override;
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static OSStatus RecordProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames,
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AudioBufferList *ioData);
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OSStatus RecordProc(AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
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UInt32 inNumberFrames, AudioBufferList *ioData);
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ALCenum open(const ALCchar *name) override;
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ALCboolean start() override;
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void stop() override;
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ALCenum captureSamples(void *buffer, ALCuint samples) override;
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ALCuint availableSamples() override;
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AudioUnit mAudioUnit{0};
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ALuint mFrameSize{0u};
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AudioStreamBasicDescription mFormat{}; // This is the OpenAL format as a CoreAudio ASBD
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SampleConverterPtr mConverter;
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RingBufferPtr mRing{nullptr};
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static constexpr inline const char *CurrentPrefix() noexcept { return "CoreAudioCapture::"; }
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DEF_NEWDEL(CoreAudioCapture)
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};
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CoreAudioCapture::~CoreAudioCapture()
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{
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if(mAudioUnit)
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AudioComponentInstanceDispose(mAudioUnit);
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mAudioUnit = 0;
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}
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OSStatus CoreAudioCapture::RecordProcC(void *inRefCon,
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AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
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{
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return static_cast<CoreAudioCapture*>(inRefCon)->RecordProc(ioActionFlags, inTimeStamp,
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inBusNumber, inNumberFrames, ioData);
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}
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OSStatus CoreAudioCapture::RecordProc(AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
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const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber), UInt32 inNumberFrames,
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AudioBufferList* UNUSED(ioData))
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{
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AudioUnitRenderActionFlags flags = 0;
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union {
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ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)*2];
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AudioBufferList list;
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} audiobuf = { { 0 } };
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auto rec_vec = mRing->getWriteVector();
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inNumberFrames = minz(inNumberFrames, rec_vec.first.len+rec_vec.second.len);
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// Fill the ringbuffer's two segments with data from the input device
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if(rec_vec.first.len >= inNumberFrames)
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{
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audiobuf.list.mNumberBuffers = 1;
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audiobuf.list.mBuffers[0].mNumberChannels = mFormat.mChannelsPerFrame;
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audiobuf.list.mBuffers[0].mData = rec_vec.first.buf;
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audiobuf.list.mBuffers[0].mDataByteSize = inNumberFrames * mFormat.mBytesPerFrame;
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}
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else
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{
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const size_t remaining{inNumberFrames-rec_vec.first.len};
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audiobuf.list.mNumberBuffers = 2;
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audiobuf.list.mBuffers[0].mNumberChannels = mFormat.mChannelsPerFrame;
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audiobuf.list.mBuffers[0].mData = rec_vec.first.buf;
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audiobuf.list.mBuffers[0].mDataByteSize = rec_vec.first.len * mFormat.mBytesPerFrame;
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audiobuf.list.mBuffers[1].mNumberChannels = mFormat.mChannelsPerFrame;
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audiobuf.list.mBuffers[1].mData = rec_vec.second.buf;
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audiobuf.list.mBuffers[1].mDataByteSize = remaining * mFormat.mBytesPerFrame;
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}
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OSStatus err{AudioUnitRender(mAudioUnit, &flags, inTimeStamp, audiobuf.list.mNumberBuffers,
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inNumberFrames, &audiobuf.list)};
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if(err != noErr)
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{
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ERR("AudioUnitRender error: %d\n", err);
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return err;
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}
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mRing->writeAdvance(inNumberFrames);
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return noErr;
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}
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ALCenum CoreAudioCapture::open(const ALCchar *name)
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{
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AudioStreamBasicDescription requestedFormat; // The application requested format
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AudioStreamBasicDescription hardwareFormat; // The hardware format
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AudioStreamBasicDescription outputFormat; // The AudioUnit output format
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AURenderCallbackStruct input;
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AudioComponentDescription desc;
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UInt32 outputFrameCount;
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UInt32 propertySize;
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AudioObjectPropertyAddress propertyAddress;
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UInt32 enableIO;
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AudioComponent comp;
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OSStatus err;
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if(!name)
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name = ca_device;
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else if(strcmp(name, ca_device) != 0)
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return ALC_INVALID_VALUE;
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desc.componentType = kAudioUnitType_Output;
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#if TARGET_OS_IOS
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desc.componentSubType = kAudioUnitSubType_RemoteIO;
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#else
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desc.componentSubType = kAudioUnitSubType_HALOutput;
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#endif
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desc.componentManufacturer = kAudioUnitManufacturer_Apple;
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desc.componentFlags = 0;
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desc.componentFlagsMask = 0;
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// Search for component with given description
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comp = AudioComponentFindNext(NULL, &desc);
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if(comp == NULL)
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{
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ERR("AudioComponentFindNext failed\n");
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return ALC_INVALID_VALUE;
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}
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// Open the component
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err = AudioComponentInstanceNew(comp, &mAudioUnit);
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if(err != noErr)
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{
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ERR("AudioComponentInstanceNew failed\n");
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return ALC_INVALID_VALUE;
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}
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// Turn off AudioUnit output
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enableIO = 0;
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err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO,
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kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_INVALID_VALUE;
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}
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// Turn on AudioUnit input
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enableIO = 1;
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err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO,
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kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
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if(err != noErr)
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{
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ERR("AudioUnitSetProperty failed\n");
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return ALC_INVALID_VALUE;
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}
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#if !TARGET_OS_IOS
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{
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// Get the default input device
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AudioDeviceID inputDevice = kAudioDeviceUnknown;
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propertySize = sizeof(AudioDeviceID);
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propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
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propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
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propertyAddress.mElement = kAudioObjectPropertyElementMaster;
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err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
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if(err != noErr)
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{
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ERR("AudioObjectGetPropertyData failed\n");
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return ALC_INVALID_VALUE;
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}
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if(inputDevice == kAudioDeviceUnknown)
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{
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ERR("No input device found\n");
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return ALC_INVALID_VALUE;
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}
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// Track the input device
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err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_CurrentDevice,
|
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kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
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if(err != noErr)
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{
|
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ERR("AudioUnitSetProperty failed\n");
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return ALC_INVALID_VALUE;
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}
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}
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#endif
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|
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// set capture callback
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input.inputProc = CoreAudioCapture::RecordProcC;
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input.inputProcRefCon = this;
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err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_SetInputCallback,
|
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kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
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if(err != noErr)
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{
|
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ERR("AudioUnitSetProperty failed\n");
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return ALC_INVALID_VALUE;
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}
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|
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// Initialize the device
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err = AudioUnitInitialize(mAudioUnit);
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if(err != noErr)
|
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{
|
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ERR("AudioUnitInitialize failed\n");
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return ALC_INVALID_VALUE;
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}
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|
|
// Get the hardware format
|
|
propertySize = sizeof(AudioStreamBasicDescription);
|
|
err = AudioUnitGetProperty(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
|
|
1, &hardwareFormat, &propertySize);
|
|
if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
|
|
{
|
|
ERR("AudioUnitGetProperty failed\n");
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
// 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:
|
|
ERR("%s samples not supported\n", DevFmtTypeString(mDevice->FmtType));
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
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:
|
|
ERR("%s not supported\n", DevFmtChannelsString(mDevice->FmtChans));
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
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, (void*)&outputFormat, sizeof(outputFormat));
|
|
if(err != noErr)
|
|
{
|
|
ERR("AudioUnitSetProperty failed\n");
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
// Set the AudioUnit output format frame count
|
|
uint64_t FrameCount64{mDevice->UpdateSize};
|
|
FrameCount64 = (FrameCount64*outputFormat.mSampleRate + mDevice->Frequency-1) /
|
|
mDevice->Frequency;
|
|
FrameCount64 += MAX_RESAMPLE_PADDING*2;
|
|
if(FrameCount64 > std::numeric_limits<uint32_t>::max()/2)
|
|
{
|
|
ERR("FrameCount too large\n");
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
outputFrameCount = static_cast<uint32_t>(FrameCount64);
|
|
err = AudioUnitSetProperty(mAudioUnit, kAudioUnitProperty_MaximumFramesPerSlice,
|
|
kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
|
|
if(err != noErr)
|
|
{
|
|
ERR("AudioUnitSetProperty failed: %d\n", err);
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
// Set up sample converter if needed
|
|
if(outputFormat.mSampleRate != mDevice->Frequency)
|
|
mConverter = CreateSampleConverter(mDevice->FmtType, mDevice->FmtType,
|
|
mFormat.mChannelsPerFrame, hardwareFormat.mSampleRate, mDevice->Frequency,
|
|
BSinc24Resampler);
|
|
|
|
mRing = CreateRingBuffer(outputFrameCount, mFrameSize, false);
|
|
if(!mRing) return ALC_INVALID_VALUE;
|
|
|
|
mDevice->DeviceName = name;
|
|
return ALC_NO_ERROR;
|
|
}
|
|
|
|
|
|
ALCboolean CoreAudioCapture::start()
|
|
{
|
|
OSStatus err{AudioOutputUnitStart(mAudioUnit)};
|
|
if(err != noErr)
|
|
{
|
|
ERR("AudioOutputUnitStart failed\n");
|
|
return ALC_FALSE;
|
|
}
|
|
return ALC_TRUE;
|
|
}
|
|
|
|
void CoreAudioCapture::stop()
|
|
{
|
|
OSStatus err{AudioOutputUnitStop(mAudioUnit)};
|
|
if(err != noErr)
|
|
ERR("AudioOutputUnitStop failed\n");
|
|
}
|
|
|
|
ALCenum CoreAudioCapture::captureSamples(void *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<ALsizei>(rec_vec.first.len);
|
|
auto got = static_cast<ALuint>(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<ALsizei>(rec_vec.second.len);
|
|
got += static_cast<ALuint>(mConverter->convert(&src1, &src1len,
|
|
static_cast<char*>(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 mRing->readSpace();
|
|
return mConverter->availableOut(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;
|
|
}
|