diff options
Diffstat (limited to 'Alc/backends/coreaudio.c')
-rw-r--r-- | Alc/backends/coreaudio.c | 816 |
1 files changed, 0 insertions, 816 deletions
diff --git a/Alc/backends/coreaudio.c b/Alc/backends/coreaudio.c deleted file mode 100644 index adb01fa6..00000000 --- a/Alc/backends/coreaudio.c +++ /dev/null @@ -1,816 +0,0 @@ -/** - * OpenAL cross platform audio library - * Copyright (C) 1999-2007 by authors. - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - * Or go to http://www.gnu.org/copyleft/lgpl.html - */ - -#include "config.h" - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> - -#include "alMain.h" -#include "alu.h" -#include "ringbuffer.h" - -#include <unistd.h> -#include <AudioUnit/AudioUnit.h> -#include <AudioToolbox/AudioToolbox.h> - -#include "backends/base.h" - - -static const ALCchar ca_device[] = "CoreAudio Default"; - - -typedef struct ALCcoreAudioPlayback { - DERIVE_FROM_TYPE(ALCbackend); - - AudioUnit audioUnit; - - ALuint frameSize; - AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD -} ALCcoreAudioPlayback; - -static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device); -static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self); -static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name); -static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self); -static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self); -static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self); -static DECLARE_FORWARD2(ALCcoreAudioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint) -static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ALCuint, availableSamples) -static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ClockLatency, getClockLatency) -static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, lock) -static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, unlock) -DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioPlayback) - -DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback); - - -static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device) -{ - ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); - SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self); - - self->frameSize = 0; - memset(&self->format, 0, sizeof(self->format)); -} - -static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self) -{ - AudioUnitUninitialize(self->audioUnit); - AudioComponentInstanceDispose(self->audioUnit); - - ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); -} - - -static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon, - AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp), - UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData) -{ - ALCcoreAudioPlayback *self = inRefCon; - ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; - - ALCcoreAudioPlayback_lock(self); - aluMixData(device, ioData->mBuffers[0].mData, - ioData->mBuffers[0].mDataByteSize / self->frameSize); - ALCcoreAudioPlayback_unlock(self); - - return noErr; -} - - -static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name) -{ - ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; - AudioComponentDescription desc; - AudioComponent comp; - OSStatus err; - - if(!name) - name = ca_device; - else if(strcmp(name, ca_device) != 0) - return ALC_INVALID_VALUE; - - /* open the default output unit */ - desc.componentType = kAudioUnitType_Output; -#if TARGET_OS_IOS - desc.componentSubType = kAudioUnitSubType_RemoteIO; -#else - desc.componentSubType = kAudioUnitSubType_DefaultOutput; -#endif - desc.componentManufacturer = kAudioUnitManufacturer_Apple; - desc.componentFlags = 0; - desc.componentFlagsMask = 0; - - comp = AudioComponentFindNext(NULL, &desc); - if(comp == NULL) - { - ERR("AudioComponentFindNext failed\n"); - return ALC_INVALID_VALUE; - } - - err = AudioComponentInstanceNew(comp, &self->audioUnit); - if(err != noErr) - { - ERR("AudioComponentInstanceNew failed\n"); - return ALC_INVALID_VALUE; - } - - /* init and start the default audio unit... */ - err = AudioUnitInitialize(self->audioUnit); - if(err != noErr) - { - ERR("AudioUnitInitialize failed\n"); - AudioComponentInstanceDispose(self->audioUnit); - return ALC_INVALID_VALUE; - } - - alstr_copy_cstr(&device->DeviceName, name); - return ALC_NO_ERROR; -} - -static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) -{ - ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; - AudioStreamBasicDescription streamFormat; - AURenderCallbackStruct input; - OSStatus err; - UInt32 size; - - err = AudioUnitUninitialize(self->audioUnit); - if(err != noErr) - ERR("-- AudioUnitUninitialize failed.\n"); - - /* retrieve default output unit's properties (output side) */ - size = sizeof(AudioStreamBasicDescription); - err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size); - if(err != noErr || size != sizeof(AudioStreamBasicDescription)) - { - ERR("AudioUnitGetProperty failed\n"); - return ALC_FALSE; - } - -#if 0 - TRACE("Output streamFormat of default output unit -\n"); - TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket); - TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame); - TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel); - TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket); - TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame); - TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate); -#endif - - /* set default output unit's input side to match output side */ - err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - return ALC_FALSE; - } - - if(device->Frequency != streamFormat.mSampleRate) - { - device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates * - streamFormat.mSampleRate / - device->Frequency); - device->Frequency = streamFormat.mSampleRate; - } - - /* FIXME: How to tell what channels are what in the output device, and how - * to specify what we're giving? eg, 6.0 vs 5.1 */ - switch(streamFormat.mChannelsPerFrame) - { - case 1: - device->FmtChans = DevFmtMono; - break; - case 2: - device->FmtChans = DevFmtStereo; - break; - case 4: - device->FmtChans = DevFmtQuad; - break; - case 6: - device->FmtChans = DevFmtX51; - break; - case 7: - device->FmtChans = DevFmtX61; - break; - case 8: - device->FmtChans = DevFmtX71; - break; - default: - ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame); - device->FmtChans = DevFmtStereo; - streamFormat.mChannelsPerFrame = 2; - break; - } - SetDefaultWFXChannelOrder(device); - - /* use channel count and sample rate from the default output unit's current - * parameters, but reset everything else */ - streamFormat.mFramesPerPacket = 1; - streamFormat.mFormatFlags = 0; - switch(device->FmtType) - { - case DevFmtUByte: - device->FmtType = DevFmtByte; - /* fall-through */ - case DevFmtByte: - streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; - streamFormat.mBitsPerChannel = 8; - break; - case DevFmtUShort: - device->FmtType = DevFmtShort; - /* fall-through */ - case DevFmtShort: - streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; - streamFormat.mBitsPerChannel = 16; - break; - case DevFmtUInt: - device->FmtType = DevFmtInt; - /* fall-through */ - case DevFmtInt: - streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger; - streamFormat.mBitsPerChannel = 32; - break; - case DevFmtFloat: - streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat; - streamFormat.mBitsPerChannel = 32; - break; - } - streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame * - streamFormat.mBitsPerChannel / 8; - streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame; - streamFormat.mFormatID = kAudioFormatLinearPCM; - streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian | - kLinearPCMFormatFlagIsPacked; - - err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - return ALC_FALSE; - } - - /* setup callback */ - self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); - input.inputProc = ALCcoreAudioPlayback_MixerProc; - input.inputProcRefCon = self; - - err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - return ALC_FALSE; - } - - /* init the default audio unit... */ - err = AudioUnitInitialize(self->audioUnit); - if(err != noErr) - { - ERR("AudioUnitInitialize failed\n"); - return ALC_FALSE; - } - - return ALC_TRUE; -} - -static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self) -{ - OSStatus err = AudioOutputUnitStart(self->audioUnit); - if(err != noErr) - { - ERR("AudioOutputUnitStart failed\n"); - return ALC_FALSE; - } - - return ALC_TRUE; -} - -static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self) -{ - OSStatus err = AudioOutputUnitStop(self->audioUnit); - if(err != noErr) - ERR("AudioOutputUnitStop failed\n"); -} - - - - -typedef struct ALCcoreAudioCapture { - DERIVE_FROM_TYPE(ALCbackend); - - AudioUnit audioUnit; - - ALuint frameSize; - ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate - AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD - - AudioConverterRef audioConverter; // Sample rate converter if needed - AudioBufferList *bufferList; // Buffer for data coming from the input device - ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling - - ll_ringbuffer_t *ring; -} ALCcoreAudioCapture; - -static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device); -static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self); -static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name); -static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ALCboolean, reset) -static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self); -static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self); -static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples); -static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self); -static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ClockLatency, getClockLatency) -static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, lock) -static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, unlock) -DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture) - -DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture); - - -static AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize) -{ - AudioBufferList *list; - - list = calloc(1, FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize); - if(list) - { - list->mNumberBuffers = 1; - - list->mBuffers[0].mNumberChannels = channelCount; - list->mBuffers[0].mDataByteSize = byteSize; - list->mBuffers[0].mData = &list->mBuffers[1]; - } - return list; -} - -static void destroy_buffer_list(AudioBufferList *list) -{ - free(list); -} - - -static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device) -{ - ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); - SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self); - - self->audioUnit = 0; - self->audioConverter = NULL; - self->bufferList = NULL; - self->resampleBuffer = NULL; - self->ring = NULL; -} - -static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self) -{ - ll_ringbuffer_free(self->ring); - self->ring = NULL; - - free(self->resampleBuffer); - self->resampleBuffer = NULL; - - destroy_buffer_list(self->bufferList); - self->bufferList = NULL; - - if(self->audioConverter) - AudioConverterDispose(self->audioConverter); - self->audioConverter = NULL; - - if(self->audioUnit) - AudioComponentInstanceDispose(self->audioUnit); - self->audioUnit = 0; - - ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); -} - - -static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon, - AudioUnitRenderActionFlags* UNUSED(ioActionFlags), - const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber), - UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData)) -{ - ALCcoreAudioCapture *self = inRefCon; - AudioUnitRenderActionFlags flags = 0; - OSStatus err; - - // fill the bufferList with data from the input device - err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList); - if(err != noErr) - { - ERR("AudioUnitRender error: %d\n", err); - return err; - } - - ll_ringbuffer_write(self->ring, self->bufferList->mBuffers[0].mData, inNumberFrames); - - return noErr; -} - -static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter), - UInt32 *ioNumberDataPackets, AudioBufferList *ioData, - AudioStreamPacketDescription** UNUSED(outDataPacketDescription), - void *inUserData) -{ - ALCcoreAudioCapture *self = inUserData; - - // Read from the ring buffer and store temporarily in a large buffer - ll_ringbuffer_read(self->ring, self->resampleBuffer, *ioNumberDataPackets); - - // Set the input data - ioData->mNumberBuffers = 1; - ioData->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame; - ioData->mBuffers[0].mData = self->resampleBuffer; - ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->format.mBytesPerFrame; - - return noErr; -} - - -static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name) -{ - ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; - AudioStreamBasicDescription requestedFormat; // The application requested format - AudioStreamBasicDescription hardwareFormat; // The hardware format - AudioStreamBasicDescription outputFormat; // The AudioUnit output format - AURenderCallbackStruct input; - AudioComponentDescription desc; - UInt32 outputFrameCount; - UInt32 propertySize; - AudioObjectPropertyAddress propertyAddress; - UInt32 enableIO; - AudioComponent comp; - OSStatus err; - - if(!name) - name = ca_device; - else if(strcmp(name, ca_device) != 0) - return ALC_INVALID_VALUE; - - desc.componentType = kAudioUnitType_Output; -#if TARGET_OS_IOS - desc.componentSubType = kAudioUnitSubType_RemoteIO; -#else - desc.componentSubType = kAudioUnitSubType_HALOutput; -#endif - desc.componentManufacturer = kAudioUnitManufacturer_Apple; - desc.componentFlags = 0; - desc.componentFlagsMask = 0; - - // Search for component with given description - comp = AudioComponentFindNext(NULL, &desc); - if(comp == NULL) - { - ERR("AudioComponentFindNext failed\n"); - return ALC_INVALID_VALUE; - } - - // Open the component - err = AudioComponentInstanceNew(comp, &self->audioUnit); - if(err != noErr) - { - ERR("AudioComponentInstanceNew failed\n"); - goto error; - } - - // Turn off AudioUnit output - enableIO = 0; - err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - goto error; - } - - // Turn on AudioUnit input - enableIO = 1; - err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - goto error; - } - -#if !TARGET_OS_IOS - // Get the default input device - AudioDeviceID inputDevice = kAudioDeviceUnknown; - - propertySize = sizeof(AudioDeviceID); - propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; - propertyAddress.mScope = kAudioObjectPropertyScopeGlobal; - propertyAddress.mElement = kAudioObjectPropertyElementMaster; - - err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice); - if(err != noErr) - { - ERR("AudioObjectGetPropertyData failed\n"); - goto error; - } - if(inputDevice == kAudioDeviceUnknown) - { - ERR("No input device found\n"); - goto error; - } - - // Track the input device - err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - goto error; - } -#endif - - // set capture callback - input.inputProc = ALCcoreAudioCapture_RecordProc; - input.inputProcRefCon = self; - - err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - goto error; - } - - // Initialize the device - err = AudioUnitInitialize(self->audioUnit); - if(err != noErr) - { - ERR("AudioUnitInitialize failed\n"); - goto error; - } - - // Get the hardware format - propertySize = sizeof(AudioStreamBasicDescription); - err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize); - if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription)) - { - ERR("AudioUnitGetProperty failed\n"); - goto error; - } - - // Set up the requested format description - switch(device->FmtType) - { - case DevFmtUByte: - requestedFormat.mBitsPerChannel = 8; - requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked; - break; - case DevFmtShort: - requestedFormat.mBitsPerChannel = 16; - requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked; - break; - case DevFmtInt: - requestedFormat.mBitsPerChannel = 32; - requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked; - break; - case DevFmtFloat: - requestedFormat.mBitsPerChannel = 32; - requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked; - break; - case DevFmtByte: - case DevFmtUShort: - case DevFmtUInt: - ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType)); - goto error; - } - - switch(device->FmtChans) - { - case DevFmtMono: - requestedFormat.mChannelsPerFrame = 1; - break; - case DevFmtStereo: - requestedFormat.mChannelsPerFrame = 2; - break; - - case DevFmtQuad: - case DevFmtX51: - case DevFmtX51Rear: - case DevFmtX61: - case DevFmtX71: - case DevFmtAmbi3D: - ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans)); - goto error; - } - - requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8; - requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame; - requestedFormat.mSampleRate = device->Frequency; - requestedFormat.mFormatID = kAudioFormatLinearPCM; - requestedFormat.mReserved = 0; - requestedFormat.mFramesPerPacket = 1; - - // save requested format description for later use - self->format = requestedFormat; - self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); - - // Use intermediate format for sample rate conversion (outputFormat) - // Set sample rate to the same as hardware for resampling later - outputFormat = requestedFormat; - outputFormat.mSampleRate = hardwareFormat.mSampleRate; - - // Determine sample rate ratio for resampling - self->sampleRateRatio = outputFormat.mSampleRate / device->Frequency; - - // The output format should be the requested format, but using the hardware sample rate - // This is because the AudioUnit will automatically scale other properties, except for sample rate - err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed\n"); - goto error; - } - - // Set the AudioUnit output format frame count - outputFrameCount = device->UpdateSize * self->sampleRateRatio; - err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount)); - if(err != noErr) - { - ERR("AudioUnitSetProperty failed: %d\n", err); - goto error; - } - - // Set up sample converter - err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter); - if(err != noErr) - { - ERR("AudioConverterNew failed: %d\n", err); - goto error; - } - - // Create a buffer for use in the resample callback - self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio); - - // Allocate buffer for the AudioUnit output - self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio); - if(self->bufferList == NULL) - goto error; - - self->ring = ll_ringbuffer_create( - (size_t)ceil(device->UpdateSize*self->sampleRateRatio*device->NumUpdates), - self->frameSize, false - ); - if(!self->ring) goto error; - - alstr_copy_cstr(&device->DeviceName, name); - - return ALC_NO_ERROR; - -error: - ll_ringbuffer_free(self->ring); - self->ring = NULL; - free(self->resampleBuffer); - self->resampleBuffer = NULL; - destroy_buffer_list(self->bufferList); - self->bufferList = NULL; - - if(self->audioConverter) - AudioConverterDispose(self->audioConverter); - self->audioConverter = NULL; - if(self->audioUnit) - AudioComponentInstanceDispose(self->audioUnit); - self->audioUnit = 0; - - return ALC_INVALID_VALUE; -} - - -static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self) -{ - OSStatus err = AudioOutputUnitStart(self->audioUnit); - if(err != noErr) - { - ERR("AudioOutputUnitStart failed\n"); - return ALC_FALSE; - } - return ALC_TRUE; -} - -static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self) -{ - OSStatus err = AudioOutputUnitStop(self->audioUnit); - if(err != noErr) - ERR("AudioOutputUnitStop failed\n"); -} - -static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples) -{ - union { - ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)]; - AudioBufferList list; - } audiobuf = { { 0 } }; - UInt32 frameCount; - OSStatus err; - - // If no samples are requested, just return - if(samples == 0) return ALC_NO_ERROR; - - // Point the resampling buffer to the capture buffer - audiobuf.list.mNumberBuffers = 1; - audiobuf.list.mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame; - audiobuf.list.mBuffers[0].mDataByteSize = samples * self->frameSize; - audiobuf.list.mBuffers[0].mData = buffer; - - // Resample into another AudioBufferList - frameCount = samples; - err = AudioConverterFillComplexBuffer(self->audioConverter, - ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL - ); - if(err != noErr) - { - ERR("AudioConverterFillComplexBuffer error: %d\n", err); - return ALC_INVALID_VALUE; - } - return ALC_NO_ERROR; -} - -static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self) -{ - return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio; -} - - -typedef struct ALCcoreAudioBackendFactory { - DERIVE_FROM_TYPE(ALCbackendFactory); -} ALCcoreAudioBackendFactory; -#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } } - -ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void); - -static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory *self); -static DECLARE_FORWARD(ALCcoreAudioBackendFactory, ALCbackendFactory, void, deinit) -static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory *self, ALCbackend_Type type); -static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory *self, enum DevProbe type, al_string *outnames); -static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory *self, ALCdevice *device, ALCbackend_Type type); -DEFINE_ALCBACKENDFACTORY_VTABLE(ALCcoreAudioBackendFactory); - - -ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void) -{ - static ALCcoreAudioBackendFactory factory = ALCCOREAUDIOBACKENDFACTORY_INITIALIZER; - return STATIC_CAST(ALCbackendFactory, &factory); -} - - -static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self)) -{ - return ALC_TRUE; -} - -static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type) -{ - if(type == ALCbackend_Playback || ALCbackend_Capture) - return ALC_TRUE; - return ALC_FALSE; -} - -static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames) -{ - switch(type) - { - case ALL_DEVICE_PROBE: - case CAPTURE_DEVICE_PROBE: - alstr_append_range(outnames, ca_device, ca_device+sizeof(ca_device)); - break; - } -} - -static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) -{ - if(type == ALCbackend_Playback) - { - ALCcoreAudioPlayback *backend; - NEW_OBJ(backend, ALCcoreAudioPlayback)(device); - if(!backend) return NULL; - return STATIC_CAST(ALCbackend, backend); - } - if(type == ALCbackend_Capture) - { - ALCcoreAudioCapture *backend; - NEW_OBJ(backend, ALCcoreAudioCapture)(device); - if(!backend) return NULL; - return STATIC_CAST(ALCbackend, backend); - } - - return NULL; -} |