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authorChris Robinson <[email protected]>2018-03-23 11:58:11 -0700
committerChris Robinson <[email protected]>2018-03-23 11:58:11 -0700
commite37634e908c7563e6281997d50800a2f9633d8d2 (patch)
tree7c0acbe616048841a48b2aa00415697a72c74922 /Alc/backends
parent63c9d95b845c0554b59eb7edf6bb05e5650633ba (diff)
Remove an unused struct and move some functions to where they're used
Diffstat (limited to 'Alc/backends')
-rw-r--r--Alc/backends/coreaudio.c62
1 files changed, 24 insertions, 38 deletions
diff --git a/Alc/backends/coreaudio.c b/Alc/backends/coreaudio.c
index d6d32cc8..eeb19dab 100644
--- a/Alc/backends/coreaudio.c
+++ b/Alc/backends/coreaudio.c
@@ -36,45 +36,9 @@
#include "backends/base.h"
-typedef struct {
- 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;
-} ca_data;
-
static const ALCchar ca_device[] = "CoreAudio Default";
-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);
-}
-
-
typedef struct ALCcoreAudioPlayback {
DERIVE_FROM_TYPE(ALCbackend);
@@ -125,10 +89,10 @@ static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
ALCcoreAudioPlayback *self = inRefCon;
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
- ALCdevice_Lock(device);
+ ALCcoreAudioPlayback_lock(self);
aluMixData(device, ioData->mBuffers[0].mData,
ioData->mBuffers[0].mDataByteSize / self->frameSize);
- ALCdevice_Unlock(device);
+ ALCcoreAudioPlayback_unlock(self);
return noErr;
}
@@ -381,6 +345,28 @@ 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);