/** * 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 #include #include #include "alMain.h" #include "alu.h" #include "ringbuffer.h" #include "converter.h" #include "backends/base.h" #include #include #include namespace { static const ALCchar ca_device[] = "CoreAudio Default"; struct CoreAudioPlayback final : public BackendBase { CoreAudioPlayback(ALCdevice *device) noexcept : BackendBase{device} { } ~CoreAudioPlayback() override; static OSStatus MixerProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData); OSStatus MixerProc(AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData); ALCenum open(const ALCchar *name) override; ALCboolean reset() override; ALCboolean 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::MixerProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { return static_cast(inRefCon)->MixerProc(ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, ioData); } OSStatus CoreAudioPlayback::MixerProc(AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp), UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData) { lock(); aluMixData(mDevice, ioData->mBuffers[0].mData, ioData->mBuffers[0].mDataByteSize/mFrameSize); unlock(); return noErr; } ALCenum CoreAudioPlayback::open(const ALCchar *name) { if(!name) name = ca_device; else if(strcmp(name, ca_device) != 0) return ALC_INVALID_VALUE; /* 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) { ERR("AudioComponentFindNext failed\n"); return ALC_INVALID_VALUE; } OSStatus err{AudioComponentInstanceNew(comp, &mAudioUnit)}; if(err != noErr) { ERR("AudioComponentInstanceNew failed\n"); return ALC_INVALID_VALUE; } /* init and start the default audio unit... */ err = AudioUnitInitialize(mAudioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); AudioComponentInstanceDispose(mAudioUnit); return ALC_INVALID_VALUE; } mDevice->DeviceName = name; return ALC_NO_ERROR; } ALCboolean 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(sizeof(AudioStreamBasicDescription)); err = AudioUnitGetProperty(mAudioUnit, 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(mAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_FALSE; } if(mDevice->Frequency != streamFormat.mSampleRate) { mDevice->BufferSize = static_cast(uint64_t{mDevice->BufferSize} * streamFormat.mSampleRate / mDevice->Frequency); mDevice->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: 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 ALC_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 ALC_FALSE; } /* init the default audio unit... */ err = AudioUnitInitialize(mAudioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); return ALC_FALSE; } return ALC_TRUE; } ALCboolean CoreAudioPlayback::start() { OSStatus err{AudioOutputUnitStart(mAudioUnit)}; if(err != noErr) { ERR("AudioOutputUnitStart failed\n"); return ALC_FALSE; } return ALC_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; static OSStatus RecordProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData); OSStatus RecordProc(AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData); ALCenum open(const ALCchar *name) override; ALCboolean start() override; void stop() override; ALCenum captureSamples(void *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::RecordProcC(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { return static_cast(inRefCon)->RecordProc(ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, ioData); } OSStatus CoreAudioCapture::RecordProc(AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber), UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData)) { AudioUnitRenderActionFlags flags = 0; union { ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)*2]; AudioBufferList list; } audiobuf = { { 0 } }; auto rec_vec = mRing->getWriteVector(); inNumberFrames = 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 size_t remaining{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 = 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; } ALCenum 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; 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, &mAudioUnit); if(err != noErr) { ERR("AudioComponentInstanceNew failed\n"); return ALC_INVALID_VALUE; } // Turn off AudioUnit output enableIO = 0; err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_INVALID_VALUE; } // Turn on AudioUnit input enableIO = 1; err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_INVALID_VALUE; } #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"); return ALC_INVALID_VALUE; } if(inputDevice == kAudioDeviceUnknown) { ERR("No input device found\n"); return ALC_INVALID_VALUE; } // Track the input device err = AudioUnitSetProperty(mAudioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); return ALC_INVALID_VALUE; } } #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) { ERR("AudioUnitSetProperty failed\n"); return ALC_INVALID_VALUE; } // Initialize the device err = AudioUnitInitialize(mAudioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); return ALC_INVALID_VALUE; } // 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::max()/2) { ERR("FrameCount too large\n"); return ALC_INVALID_VALUE; } outputFrameCount = static_cast(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(rec_vec.first.len); auto got = static_cast(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(rec_vec.second.len); got += static_cast(mConverter->convert(&src1, &src1len, static_cast(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; }