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Update OpenAL Soft to commit 414b56edec5441211dc924fef365c54267c04f1c

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This commit is contained in:
Alex Szpakowski
2018-04-03 20:46:26 -03:00
parent ac70ec9f6f
commit 5be7c968b8
138 changed files with 18084 additions and 18482 deletions
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libs/openal-soft/Alc/backends/coreaudio.c
+66 -84
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@@ -23,10 +23,10 @@
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <alloca.h>
#include "alMain.h"
#include "alu.h"
#include "ringbuffer.h"
#include <CoreServices/CoreServices.h>
#include <unistd.h>
@@ -36,56 +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, sizeof(AudioBufferList) + sizeof(AudioBuffer));
if(list)
{
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = channelCount;
list->mBuffers[0].mDataByteSize = byteSize;
list->mBuffers[0].mData = malloc(byteSize);
if(list->mBuffers[0].mData == NULL)
{
free(list);
list = NULL;
}
}
return list;
}
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);
}
}
typedef struct ALCcoreAudioPlayback {
DERIVE_FROM_TYPE(ALCbackend);
@@ -98,7 +51,6 @@ typedef struct 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);
@@ -123,6 +75,9 @@ static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice
static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
{
AudioUnitUninitialize(self->audioUnit);
AudioComponentInstanceDispose(self->audioUnit);
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
@@ -134,10 +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;
}
@@ -189,12 +144,6 @@ static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCch
return ALC_NO_ERROR;
}
static void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self)
{
AudioUnitUninitialize(self->audioUnit);
AudioComponentInstanceDispose(self->audioUnit);
}
static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
@@ -382,7 +331,6 @@ typedef struct 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);
@@ -396,15 +344,59 @@ 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));
}
@@ -667,8 +659,8 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar
goto error;
self->ring = ll_ringbuffer_create(
device->UpdateSize*self->sampleRateRatio*device->NumUpdates + 1,
self->frameSize
(size_t)ceil(device->UpdateSize*self->sampleRateRatio*device->NumUpdates),
self->frameSize, false
);
if(!self->ring) goto error;
@@ -680,30 +672,21 @@ 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 void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self)
{
ll_ringbuffer_free(self->ring);
self->ring = NULL;
free(self->resampleBuffer);
destroy_buffer_list(self->bufferList);
AudioConverterDispose(self->audioConverter);
AudioComponentInstanceDispose(self->audioUnit);
}
static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
{
OSStatus err = AudioOutputUnitStart(self->audioUnit);
@@ -724,27 +707,26 @@ static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
{
AudioBufferList *list;
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;
// Allocate a temporary AudioBufferList to use as the return resamples data
list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
if(samples == 0) return ALC_NO_ERROR;
// Point the resampling buffer to the capture buffer
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
list->mBuffers[0].mDataByteSize = samples * self->frameSize;
list->mBuffers[0].mData = 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, list, NULL
ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL
);
if(err != noErr)
{