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authorChris Robinson <[email protected]>2017-04-09 11:21:02 -0700
committerChris Robinson <[email protected]>2017-04-09 11:21:02 -0700
commit81527cdbddc52338f5fb3c8b79139bf9d9186d3a (patch)
tree17cc0f2e175d40e3d1fd9e29564560b1d60b3f8c /Alc/backends
parentaef774a7a03485a645edc79ea292abe8c634b8e9 (diff)
Convert the CoreAudio backend to the updated backend API
Diffstat (limited to 'Alc/backends')
-rw-r--r--Alc/backends/base.h1
-rw-r--r--Alc/backends/coreaudio.c460
2 files changed, 283 insertions, 178 deletions
diff --git a/Alc/backends/base.h b/Alc/backends/base.h
index 4f398047..961a4d1a 100644
--- a/Alc/backends/base.h
+++ b/Alc/backends/base.h
@@ -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);
diff --git a/Alc/backends/coreaudio.c b/Alc/backends/coreaudio.c
index 5e0b03bd..435c0fae 100644
--- a/Alc/backends/coreaudio.c
+++ b/Alc/backends/coreaudio.c
@@ -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;
-
- ALCdevice_Lock(device);
- aluMixData(device, ioData->mBuffers[0].mData,
- ioData->mBuffers[0].mDataByteSize / data->frameSize);
- ALCdevice_Unlock(device);
-
- return noErr;
+ if(list)
+ {
+ UInt32 i;
+ for(i = 0;i < list->mNumberBuffers;i++)
+ free(list->mBuffers[i].mData);
+ free(list);
+ }
}
-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);
+typedef struct ALCcoreAudioPlayback {
+ DERIVE_FROM_TYPE(ALCbackend);
- // 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;
+ AudioUnit audioUnit;
- return noErr;
+ 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 OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
- const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
- UInt32 inNumberFrames, AudioBufferList *ioData)
+static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
{
- ALCdevice *device = (ALCdevice*)inRefCon;
- ca_data *data = (ca_data*)device->ExtraData;
- AudioUnitRenderActionFlags flags = 0;
- OSStatus err;
+ ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
+}
- // 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);
+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 / self->frameSize);
+ ALCdevice_Unlock(device);
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 = 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 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;
- err = AudioOutputUnitStop(data->audioUnit);
+ // fill the bufferList with data from the input device
+ err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList);
if(err != noErr)
- ERR("AudioOutputUnitStop failed\n");
+ {
+ 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 ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
+
+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);
-
- if(data->audioConverter)
- AudioConverterDispose(data->audioConverter);
- if(data->audioUnit)
- AudioComponentInstanceDispose(data->audioUnit);
+ ll_ringbuffer_free(self->ring);
+ self->ring = NULL;
+ free(self->resampleBuffer);
+ destroy_buffer_list(self->bufferList);
- 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
-};
-
-ALCboolean alc_ca_init(BackendFuncs *func_list)
+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))
{
- *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;
+}