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/**
* OpenAL cross platform audio library
* Copyright (C) 2018 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 <stdlib.h>
#include <SDL2/SDL.h>
#include "alMain.h"
#include "alu.h"
#include "threads.h"
#include "compat.h"
#include "backends/base.h"
#ifdef _WIN32
#define DEVNAME_PREFIX "OpenAL Soft on "
#else
#define DEVNAME_PREFIX ""
#endif
typedef struct ALCsdl2Backend {
DERIVE_FROM_TYPE(ALCbackend);
SDL_AudioDeviceID deviceID;
ALsizei frameSize;
ALuint Frequency;
enum DevFmtChannels FmtChans;
enum DevFmtType FmtType;
ALuint UpdateSize;
} ALCsdl2Backend;
static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device);
static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self);
static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name);
static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self);
static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self);
static void ALCsdl2Backend_stop(ALCsdl2Backend *self);
static DECLARE_FORWARD2(ALCsdl2Backend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ClockLatency, getClockLatency)
static void ALCsdl2Backend_lock(ALCsdl2Backend *self);
static void ALCsdl2Backend_unlock(ALCsdl2Backend *self);
DECLARE_DEFAULT_ALLOCATORS(ALCsdl2Backend)
DEFINE_ALCBACKEND_VTABLE(ALCsdl2Backend);
static const ALCchar defaultDeviceName[] = DEVNAME_PREFIX "Default Device";
static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCsdl2Backend, ALCbackend, self);
self->deviceID = 0;
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
self->Frequency = device->Frequency;
self->FmtChans = device->FmtChans;
self->FmtType = device->FmtType;
self->UpdateSize = device->UpdateSize;
}
static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self)
{
if(self->deviceID)
SDL_CloseAudioDevice(self->deviceID);
self->deviceID = 0;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static void ALCsdl2Backend_audioCallback(void *ptr, Uint8 *stream, int len)
{
ALCsdl2Backend *self = (ALCsdl2Backend*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
assert((len % self->frameSize) == 0);
aluMixData(device, stream, len / self->frameSize);
}
static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
SDL_AudioSpec want, have;
SDL_zero(want);
SDL_zero(have);
want.freq = device->Frequency;
switch(device->FmtType)
{
case DevFmtUByte: want.format = AUDIO_U8; break;
case DevFmtByte: want.format = AUDIO_S8; break;
case DevFmtUShort: want.format = AUDIO_U16SYS; break;
case DevFmtShort: want.format = AUDIO_S16SYS; break;
case DevFmtUInt: /* fall-through */
case DevFmtInt: want.format = AUDIO_S32SYS; break;
case DevFmtFloat: want.format = AUDIO_F32; break;
}
want.channels = (device->FmtChans == DevFmtMono) ? 1 : 2;
want.samples = device->UpdateSize;
want.callback = ALCsdl2Backend_audioCallback;
want.userdata = self;
/* Passing NULL to SDL_OpenAudioDevice opens a default, which isn't
* necessarily the first in the list.
*/
if(!name || strcmp(name, defaultDeviceName) == 0)
self->deviceID = SDL_OpenAudioDevice(NULL, SDL_FALSE, &want, &have,
SDL_AUDIO_ALLOW_ANY_CHANGE);
else
{
const size_t prefix_len = strlen(DEVNAME_PREFIX);
if(strncmp(name, DEVNAME_PREFIX, prefix_len) == 0)
self->deviceID = SDL_OpenAudioDevice(name+prefix_len, SDL_FALSE, &want, &have,
SDL_AUDIO_ALLOW_ANY_CHANGE);
else
self->deviceID = SDL_OpenAudioDevice(name, SDL_FALSE, &want, &have,
SDL_AUDIO_ALLOW_ANY_CHANGE);
}
if(self->deviceID == 0)
return ALC_INVALID_VALUE;
device->Frequency = have.freq;
if(have.channels == 1)
device->FmtChans = DevFmtMono;
else if(have.channels == 2)
device->FmtChans = DevFmtStereo;
else
{
ERR("Got unhandled SDL channel count: %d\n", (int)have.channels);
return ALC_INVALID_VALUE;
}
switch(have.format)
{
case AUDIO_U8: device->FmtType = DevFmtUByte; break;
case AUDIO_S8: device->FmtType = DevFmtByte; break;
case AUDIO_U16SYS: device->FmtType = DevFmtUShort; break;
case AUDIO_S16SYS: device->FmtType = DevFmtShort; break;
case AUDIO_S32SYS: device->FmtType = DevFmtInt; break;
case AUDIO_F32SYS: device->FmtType = DevFmtFloat; break;
default:
ERR("Got unsupported SDL format: 0x%04x\n", have.format);
return ALC_INVALID_VALUE;
}
device->UpdateSize = have.samples;
device->NumUpdates = 2; /* SDL always (tries to) use two periods. */
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
self->Frequency = device->Frequency;
self->FmtChans = device->FmtChans;
self->FmtType = device->FmtType;
self->UpdateSize = device->UpdateSize;
alstr_copy_cstr(&device->DeviceName, name ? name : defaultDeviceName);
return ALC_NO_ERROR;
}
static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
device->Frequency = self->Frequency;
device->FmtChans = self->FmtChans;
device->FmtType = self->FmtType;
device->UpdateSize = self->UpdateSize;
device->NumUpdates = 2;
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self)
{
SDL_PauseAudioDevice(self->deviceID, 0);
return ALC_TRUE;
}
static void ALCsdl2Backend_stop(ALCsdl2Backend *self)
{
SDL_PauseAudioDevice(self->deviceID, 1);
}
static void ALCsdl2Backend_lock(ALCsdl2Backend *self)
{
SDL_LockAudioDevice(self->deviceID);
}
static void ALCsdl2Backend_unlock(ALCsdl2Backend *self)
{
SDL_UnlockAudioDevice(self->deviceID);
}
typedef struct ALCsdl2BackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCsdl2BackendFactory;
#define ALCsdl2BACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsdl2BackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void);
static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory *self);
static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory *self);
static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory *self, ALCbackend_Type type);
static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsdl2BackendFactory);
ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void)
{
static ALCsdl2BackendFactory factory = ALCsdl2BACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory* UNUSED(self))
{
if(SDL_InitSubSystem(SDL_INIT_AUDIO) == 0)
return AL_TRUE;
return ALC_FALSE;
}
static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory* UNUSED(self))
{
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
int num_devices, i;
al_string name;
if(type != ALL_DEVICE_PROBE)
return;
AL_STRING_INIT(name);
num_devices = SDL_GetNumAudioDevices(SDL_FALSE);
alstr_append_range(outnames, defaultDeviceName, defaultDeviceName+sizeof(defaultDeviceName));
for(i = 0;i < num_devices;++i)
{
alstr_copy_cstr(&name, DEVNAME_PREFIX);
alstr_append_cstr(&name, SDL_GetAudioDeviceName(i, SDL_FALSE));
if(!alstr_empty(name))
alstr_append_range(outnames, VECTOR_BEGIN(name), VECTOR_END(name)+1);
}
alstr_reset(&name);
}
static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCsdl2Backend *backend;
NEW_OBJ(backend, ALCsdl2Backend)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
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