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-rw-r--r--Alc/backends/coreaudio.c816
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;
-}