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|
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "alMain.h"
#include "alu.h"
#include "ringbuffer.h"
#include <CoreServices/CoreServices.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;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
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;
AudioDeviceID inputDevice;
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;
desc.componentSubType = kAudioUnitSubType_HALOutput;
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;
}
// Get the default input device
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;
}
// 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);
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)
{
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(ca_device);
break;
case CAPTURE_DEVICE_PROBE:
AppendCaptureDeviceList(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;
}
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