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-rw-r--r--Alc/backends/sdl2.c288
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diff --git a/Alc/backends/sdl2.c b/Alc/backends/sdl2.c
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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;
+}