Convert the CoreAudio backend to the updated backend API

This commit is contained in:
Chris Robinson 2017-04-09 11:21:02 -07:00
parent aef774a7a0
commit 81527cdbdd
4 changed files with 290 additions and 188 deletions

View File

@ -72,7 +72,7 @@ static struct BackendInfo BackendList[] = {
{ "alsa", ALCalsaBackendFactory_getFactory, NULL, NULL, NULL, EmptyFuncs },
#endif
#ifdef HAVE_COREAUDIO
{ "core", NULL, alc_ca_init, alc_ca_deinit, alc_ca_probe, EmptyFuncs },
{ "core", ALCcoreAudioBackendFactory_getFactory, NULL, NULL, NULL, EmptyFuncs },
#endif
#ifdef HAVE_OSS
{ "oss", ALCossBackendFactory_getFactory, NULL, NULL, NULL, EmptyFuncs },

View File

@ -137,6 +137,7 @@ static const struct ALCbackendFactoryVtable T##_ALCbackendFactory_vtable = { \
ALCbackendFactory *ALCpulseBackendFactory_getFactory(void);
ALCbackendFactory *ALCalsaBackendFactory_getFactory(void);
ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
ALCbackendFactory *ALCossBackendFactory_getFactory(void);
ALCbackendFactory *ALCjackBackendFactory_getFactory(void);
ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);

View File

@ -33,6 +33,8 @@
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
#include "backends/base.h"
typedef struct {
AudioUnit audioUnit;
@ -51,17 +53,6 @@ typedef struct {
static const ALCchar ca_device[] = "CoreAudio Default";
static void destroy_buffer_list(AudioBufferList* list)
{
if(list)
{
UInt32 i;
for(i = 0;i < list->mNumberBuffers;i++)
free(list->mBuffers[i].mData);
free(list);
}
}
static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
{
AudioBufferList *list;
@ -83,70 +74,85 @@ static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSiz
return list;
}
static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
static void destroy_buffer_list(AudioBufferList* list)
{
ALCdevice *device = (ALCdevice*)inRefCon;
ca_data *data = (ca_data*)device->ExtraData;
if(list)
{
UInt32 i;
for(i = 0;i < list->mNumberBuffers;i++)
free(list->mBuffers[i].mData);
free(list);
}
}
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 void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self);
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)
{
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;
ALCdevice_Lock(device);
aluMixData(device, ioData->mBuffers[0].mData,
ioData->mBuffers[0].mDataByteSize / data->frameSize);
ioData->mBuffers[0].mDataByteSize / self->frameSize);
ALCdevice_Unlock(device);
return noErr;
}
static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
{
ALCdevice *device = (ALCdevice*)inUserData;
ca_data *data = (ca_data*)device->ExtraData;
// Read from the ring buffer and store temporarily in a large buffer
ll_ringbuffer_read(data->ring, data->resampleBuffer, *ioNumberDataPackets);
// Set the input data
ioData->mNumberBuffers = 1;
ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
ioData->mBuffers[0].mData = data->resampleBuffer;
ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
return noErr;
}
static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList *ioData)
{
ALCdevice *device = (ALCdevice*)inRefCon;
ca_data *data = (ca_data*)device->ExtraData;
AudioUnitRenderActionFlags flags = 0;
OSStatus err;
// fill the bufferList with data from the input device
err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
if(err != noErr)
{
ERR("AudioUnitRender error: %d\n", err);
return err;
}
ll_ringbuffer_write(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
return noErr;
}
static ALCenum ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
AudioComponentDescription desc;
AudioComponent comp;
ca_data *data;
OSStatus err;
if(!deviceName)
deviceName = ca_device;
else if(strcmp(deviceName, ca_device) != 0)
if(!name)
name = ca_device;
else if(strcmp(name, ca_device) != 0)
return ALC_INVALID_VALUE;
/* open the default output unit */
@ -163,57 +169,47 @@ static ALCenum ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
return ALC_INVALID_VALUE;
}
data = calloc(1, sizeof(*data));
err = AudioComponentInstanceNew(comp, &data->audioUnit);
err = AudioComponentInstanceNew(comp, &self->audioUnit);
if(err != noErr)
{
ERR("AudioComponentInstanceNew failed\n");
free(data);
return ALC_INVALID_VALUE;
}
/* init and start the default audio unit... */
err = AudioUnitInitialize(data->audioUnit);
err = AudioUnitInitialize(self->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
AudioComponentInstanceDispose(data->audioUnit);
free(data);
AudioComponentInstanceDispose(self->audioUnit);
return ALC_INVALID_VALUE;
}
alstr_copy_cstr(&device->DeviceName, deviceName);
device->ExtraData = data;
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static void ca_close_playback(ALCdevice *device)
static void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self)
{
ca_data *data = (ca_data*)device->ExtraData;
AudioUnitUninitialize(data->audioUnit);
AudioComponentInstanceDispose(data->audioUnit);
free(data);
device->ExtraData = NULL;
AudioUnitUninitialize(self->audioUnit);
AudioComponentInstanceDispose(self->audioUnit);
}
static ALCboolean ca_reset_playback(ALCdevice *device)
static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
{
ca_data *data = (ca_data*)device->ExtraData;
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
AudioStreamBasicDescription streamFormat;
AURenderCallbackStruct input;
OSStatus err;
UInt32 size;
err = AudioUnitUninitialize(data->audioUnit);
err = AudioUnitUninitialize(self->audioUnit);
if(err != noErr)
ERR("-- AudioUnitUninitialize failed.\n");
/* retrieve default output unit's properties (output side) */
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
if(err != noErr || size != sizeof(AudioStreamBasicDescription))
{
ERR("AudioUnitGetProperty failed\n");
@ -231,7 +227,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
#endif
/* set default output unit's input side to match output side */
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -315,7 +311,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
kLinearPCMFormatFlagIsPacked;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -323,11 +319,11 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
}
/* setup callback */
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
input.inputProc = ca_callback;
input.inputProcRefCon = device;
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
input.inputProc = ALCcoreAudioPlayback_MixerProc;
input.inputProcRefCon = self;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -335,7 +331,7 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
}
/* init the default audio unit... */
err = AudioUnitInitialize(data->audioUnit);
err = AudioUnitInitialize(self->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
@ -345,12 +341,9 @@ static ALCboolean ca_reset_playback(ALCdevice *device)
return ALC_TRUE;
}
static ALCboolean ca_start_playback(ALCdevice *device)
static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err;
err = AudioOutputUnitStart(data->audioUnit);
OSStatus err = AudioOutputUnitStart(self->audioUnit);
if(err != noErr)
{
ERR("AudioOutputUnitStart failed\n");
@ -360,18 +353,107 @@ static ALCboolean ca_start_playback(ALCdevice *device)
return ALC_TRUE;
}
static void ca_stop_playback(ALCdevice *device)
static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err;
err = AudioOutputUnitStop(data->audioUnit);
OSStatus err = AudioOutputUnitStop(self->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
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 void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self);
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 void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self);
}
static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self)
{
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
@ -383,12 +465,11 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
AudioObjectPropertyAddress propertyAddress;
UInt32 enableIO;
AudioComponent comp;
ca_data *data;
OSStatus err;
if(!deviceName)
deviceName = ca_device;
else if(strcmp(deviceName, ca_device) != 0)
if(!name)
name = ca_device;
else if(strcmp(name, ca_device) != 0)
return ALC_INVALID_VALUE;
desc.componentType = kAudioUnitType_Output;
@ -405,11 +486,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
return ALC_INVALID_VALUE;
}
data = calloc(1, sizeof(*data));
device->ExtraData = data;
// Open the component
err = AudioComponentInstanceNew(comp, &data->audioUnit);
err = AudioComponentInstanceNew(comp, &self->audioUnit);
if(err != noErr)
{
ERR("AudioComponentInstanceNew failed\n");
@ -418,7 +496,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
// Turn off AudioUnit output
enableIO = 0;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -427,7 +505,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
// Turn on AudioUnit input
enableIO = 1;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -455,7 +533,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
}
// Track the input device
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -463,10 +541,10 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
}
// set capture callback
input.inputProc = ca_capture_callback;
input.inputProcRefCon = device;
input.inputProc = ALCcoreAudioCapture_RecordProc;
input.inputProcRefCon = self;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -474,7 +552,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
}
// Initialize the device
err = AudioUnitInitialize(data->audioUnit);
err = AudioUnitInitialize(self->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
@ -483,7 +561,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
// Get the hardware format
propertySize = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
{
ERR("AudioUnitGetProperty failed\n");
@ -545,8 +623,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
requestedFormat.mFramesPerPacket = 1;
// save requested format description for later use
data->format = requestedFormat;
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
self->format = requestedFormat;
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
// Use intermediate format for sample rate conversion (outputFormat)
// Set sample rate to the same as hardware for resampling later
@ -554,11 +632,11 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
outputFormat.mSampleRate = hardwareFormat.mSampleRate;
// Determine sample rate ratio for resampling
data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
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(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
@ -566,8 +644,8 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
}
// Set the AudioUnit output format frame count
outputFrameCount = device->UpdateSize * data->sampleRateRatio;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
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);
@ -575,7 +653,7 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
}
// Set up sample converter
err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter);
if(err != noErr)
{
ERR("AudioConverterNew failed: %d\n", err);
@ -583,75 +661,71 @@ static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
}
// Create a buffer for use in the resample callback
data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
// Allocate buffer for the AudioUnit output
data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
if(data->bufferList == NULL)
self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
if(self->bufferList == NULL)
goto error;
data->ring = ll_ringbuffer_create(
device->UpdateSize*data->sampleRateRatio*device->NumUpdates + 1,
data->frameSize
self->ring = ll_ringbuffer_create(
device->UpdateSize*self->sampleRateRatio*device->NumUpdates + 1,
self->frameSize
);
if(!data->ring) goto error;
if(!self->ring) goto error;
alstr_copy_cstr(&device->DeviceName, deviceName);
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
error:
ll_ringbuffer_free(data->ring);
data->ring = NULL;
free(data->resampleBuffer);
destroy_buffer_list(data->bufferList);
ll_ringbuffer_free(self->ring);
self->ring = NULL;
free(self->resampleBuffer);
destroy_buffer_list(self->bufferList);
if(data->audioConverter)
AudioConverterDispose(data->audioConverter);
if(data->audioUnit)
AudioComponentInstanceDispose(data->audioUnit);
free(data);
device->ExtraData = NULL;
if(self->audioConverter)
AudioConverterDispose(self->audioConverter);
if(self->audioUnit)
AudioComponentInstanceDispose(self->audioUnit);
return ALC_INVALID_VALUE;
}
static void ca_close_capture(ALCdevice *device)
static void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self)
{
ca_data *data = (ca_data*)device->ExtraData;
ll_ringbuffer_free(self->ring);
self->ring = NULL;
ll_ringbuffer_free(data->ring);
data->ring = NULL;
free(data->resampleBuffer);
destroy_buffer_list(data->bufferList);
free(self->resampleBuffer);
AudioConverterDispose(data->audioConverter);
AudioComponentInstanceDispose(data->audioUnit);
destroy_buffer_list(self->bufferList);
free(data);
device->ExtraData = NULL;
AudioConverterDispose(self->audioConverter);
AudioComponentInstanceDispose(self->audioUnit);
}
static void ca_start_capture(ALCdevice *device)
static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err = AudioOutputUnitStart(data->audioUnit);
OSStatus err = AudioOutputUnitStart(self->audioUnit);
if(err != noErr)
{
ERR("AudioOutputUnitStart failed\n");
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ca_stop_capture(ALCdevice *device)
static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err = AudioOutputUnitStop(data->audioUnit);
OSStatus err = AudioOutputUnitStop(self->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
{
ca_data *data = (ca_data*)device->ExtraData;
AudioBufferList *list;
UInt32 frameCount;
OSStatus err;
@ -665,14 +739,15 @@ static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint sa
// Point the resampling buffer to the capture buffer
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
list->mBuffers[0].mDataByteSize = samples * data->frameSize;
list->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
list->mBuffers[0].mDataByteSize = samples * self->frameSize;
list->mBuffers[0].mData = buffer;
// Resample into another AudioBufferList
frameCount = samples;
err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback,
device, &frameCount, list, NULL);
err = AudioConverterFillComplexBuffer(self->audioConverter,
ALCcoreAudioCapture_ConvertCallback, self, &frameCount, list, NULL
);
if(err != noErr)
{
ERR("AudioConverterFillComplexBuffer error: %d\n", err);
@ -681,38 +756,47 @@ static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint sa
return ALC_NO_ERROR;
}
static ALCuint ca_available_samples(ALCdevice *device)
static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
{
ca_data *data = device->ExtraData;
return ll_ringbuffer_read_space(data->ring) / data->sampleRateRatio;
return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
}
static const BackendFuncs ca_funcs = {
ca_open_playback,
ca_close_playback,
ca_reset_playback,
ca_start_playback,
ca_stop_playback,
ca_open_capture,
ca_close_capture,
ca_start_capture,
ca_stop_capture,
ca_capture_samples,
ca_available_samples
};
typedef struct ALCcoreAudioBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCcoreAudioBackendFactory;
#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } }
ALCboolean alc_ca_init(BackendFuncs *func_list)
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))
{
*func_list = ca_funcs;
return ALC_TRUE;
}
void alc_ca_deinit(void)
static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback || ALCbackend_Capture)
return ALC_TRUE;
return ALC_FALSE;
}
void alc_ca_probe(enum DevProbe type)
static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type)
{
switch(type)
{
@ -724,3 +808,23 @@ void alc_ca_probe(enum DevProbe type)
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;
}

View File

@ -429,9 +429,6 @@ typedef struct {
ALCuint (*AvailableSamples)(ALCdevice*);
} BackendFuncs;
ALCboolean alc_ca_init(BackendFuncs *func_list);
void alc_ca_deinit(void);
void alc_ca_probe(enum DevProbe type);
ALCboolean alc_qsa_init(BackendFuncs *func_list);
void alc_qsa_deinit(void);
void alc_qsa_probe(enum DevProbe type);