initial commit-moved from vulkan_guide

third_party/SDL/src/SDL.c

@@ -0,0 +1,651 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+#if defined(__WIN32__) || defined(__GDK__)
+#include "core/windows/SDL_windows.h"
+#elif defined(__OS2__)
+#include <stdlib.h> /* _exit() */
+#elif !defined(__WINRT__)
+#include <unistd.h> /* _exit(), etc. */
+#endif
+#if defined(__OS2__)
+#include "core/os2/SDL_os2.h"
+#if SDL_THREAD_OS2
+#include "thread/os2/SDL_systls_c.h"
+#endif
+#endif
+
+/* this checks for HAVE_DBUS_DBUS_H internally. */
+#include "core/linux/SDL_dbus.h"
+
+#if defined(__EMSCRIPTEN__)
+#include <emscripten.h>
+#endif
+
+/* Initialization code for SDL */
+
+#include "SDL.h"
+#include "SDL_bits.h"
+#include "SDL_revision.h"
+#include "SDL_assert_c.h"
+#include "SDL_log_c.h"
+#include "events/SDL_events_c.h"
+#include "haptic/SDL_haptic_c.h"
+#include "joystick/SDL_joystick_c.h"
+#include "sensor/SDL_sensor_c.h"
+
+/* Initialization/Cleanup routines */
+#if !SDL_TIMERS_DISABLED
+#include "timer/SDL_timer_c.h"
+#endif
+#if SDL_VIDEO_DRIVER_WINDOWS
+extern int SDL_HelperWindowCreate(void);
+extern int SDL_HelperWindowDestroy(void);
+#endif
+
+#ifdef SDL_BUILD_MAJOR_VERSION
+SDL_COMPILE_TIME_ASSERT(SDL_BUILD_MAJOR_VERSION,
+                        SDL_MAJOR_VERSION == SDL_BUILD_MAJOR_VERSION);
+SDL_COMPILE_TIME_ASSERT(SDL_BUILD_MINOR_VERSION,
+                        SDL_MINOR_VERSION == SDL_BUILD_MINOR_VERSION);
+SDL_COMPILE_TIME_ASSERT(SDL_BUILD_MICRO_VERSION,
+                        SDL_PATCHLEVEL == SDL_BUILD_MICRO_VERSION);
+#endif
+
+SDL_COMPILE_TIME_ASSERT(SDL_MAJOR_VERSION_min, SDL_MAJOR_VERSION >= 0);
+/* Limited only by the need to fit in SDL_version */
+SDL_COMPILE_TIME_ASSERT(SDL_MAJOR_VERSION_max, SDL_MAJOR_VERSION <= 255);
+
+SDL_COMPILE_TIME_ASSERT(SDL_MINOR_VERSION_min, SDL_MINOR_VERSION >= 0);
+/* Limited only by the need to fit in SDL_version */
+SDL_COMPILE_TIME_ASSERT(SDL_MINOR_VERSION_max, SDL_MINOR_VERSION <= 255);
+
+SDL_COMPILE_TIME_ASSERT(SDL_PATCHLEVEL_min, SDL_PATCHLEVEL >= 0);
+/* Limited by its encoding in SDL_VERSIONNUM and in the ABI versions */
+SDL_COMPILE_TIME_ASSERT(SDL_PATCHLEVEL_max, SDL_PATCHLEVEL <= 99);
+
+/* This is not declared in any header, although it is shared between some
+    parts of SDL, because we don't want anything calling it without an
+    extremely good reason. */
+extern SDL_NORETURN void SDL_ExitProcess(int exitcode);
+SDL_NORETURN void SDL_ExitProcess(int exitcode)
+{
+#if defined(__WIN32__) || defined(__GDK__)
+    /* "if you do not know the state of all threads in your process, it is
+       better to call TerminateProcess than ExitProcess"
+       https://msdn.microsoft.com/en-us/library/windows/desktop/ms682658(v=vs.85).aspx */
+    TerminateProcess(GetCurrentProcess(), exitcode);
+    /* MingW doesn't have TerminateProcess marked as noreturn, so add an
+       ExitProcess here that will never be reached but make MingW happy. */
+    ExitProcess(exitcode);
+#elif defined(__EMSCRIPTEN__)
+    emscripten_cancel_main_loop();   /* this should "kill" the app. */
+    emscripten_force_exit(exitcode); /* this should "kill" the app. */
+    exit(exitcode);
+#elif defined(__HAIKU__)  /* Haiku has _Exit, but it's not marked noreturn. */
+    _exit(exitcode);
+#elif defined(HAVE__EXIT) /* Upper case _Exit() */
+    _Exit(exitcode);
+#else
+    _exit(exitcode);
+#endif
+}
+
+/* The initialized subsystems */
+#ifdef SDL_MAIN_NEEDED
+static SDL_bool SDL_MainIsReady = SDL_FALSE;
+#else
+static SDL_bool SDL_MainIsReady = SDL_TRUE;
+#endif
+static SDL_bool SDL_bInMainQuit = SDL_FALSE;
+static Uint8 SDL_SubsystemRefCount[32];
+
+/* Private helper to increment a subsystem's ref counter. */
+static void SDL_PrivateSubsystemRefCountIncr(Uint32 subsystem)
+{
+    const int subsystem_index = SDL_MostSignificantBitIndex32(subsystem);
+    SDL_assert((subsystem_index < 0) || (SDL_SubsystemRefCount[subsystem_index] < 255));
+    if (subsystem_index >= 0) {
+        ++SDL_SubsystemRefCount[subsystem_index];
+    }
+}
+
+/* Private helper to decrement a subsystem's ref counter. */
+static void SDL_PrivateSubsystemRefCountDecr(Uint32 subsystem)
+{
+    const int subsystem_index = SDL_MostSignificantBitIndex32(subsystem);
+    if ((subsystem_index >= 0) && (SDL_SubsystemRefCount[subsystem_index] > 0)) {
+        --SDL_SubsystemRefCount[subsystem_index];
+    }
+}
+
+/* Private helper to check if a system needs init. */
+static SDL_bool SDL_PrivateShouldInitSubsystem(Uint32 subsystem)
+{
+    const int subsystem_index = SDL_MostSignificantBitIndex32(subsystem);
+    SDL_assert((subsystem_index < 0) || (SDL_SubsystemRefCount[subsystem_index] < 255));
+    return ((subsystem_index >= 0) && (SDL_SubsystemRefCount[subsystem_index] == 0)) ? SDL_TRUE : SDL_FALSE;
+}
+
+/* Private helper to check if a system needs to be quit. */
+static SDL_bool SDL_PrivateShouldQuitSubsystem(Uint32 subsystem)
+{
+    const int subsystem_index = SDL_MostSignificantBitIndex32(subsystem);
+    if ((subsystem_index >= 0) && (SDL_SubsystemRefCount[subsystem_index] == 0)) {
+        return SDL_FALSE;
+    }
+
+    /* If we're in SDL_Quit, we shut down every subsystem, even if refcount
+     * isn't zero.
+     */
+    return (((subsystem_index >= 0) && (SDL_SubsystemRefCount[subsystem_index] == 1)) || SDL_bInMainQuit) ? SDL_TRUE : SDL_FALSE;
+}
+
+void SDL_SetMainReady(void)
+{
+    SDL_MainIsReady = SDL_TRUE;
+}
+
+int SDL_InitSubSystem(Uint32 flags)
+{
+    Uint32 flags_initialized = 0;
+
+    if (!SDL_MainIsReady) {
+        return SDL_SetError("Application didn't initialize properly, did you include SDL_main.h in the file containing your main() function?");
+    }
+
+    SDL_LogInit();
+
+    /* Clear the error message */
+    SDL_ClearError();
+
+#if SDL_USE_LIBDBUS
+    SDL_DBus_Init();
+#endif
+
+    if (flags & SDL_INIT_GAMECONTROLLER) {
+        /* game controller implies joystick */
+        flags |= SDL_INIT_JOYSTICK;
+    }
+
+    if (flags & (SDL_INIT_VIDEO | SDL_INIT_JOYSTICK | SDL_INIT_AUDIO)) {
+        /* video or joystick or audio implies events */
+        flags |= SDL_INIT_EVENTS;
+    }
+
+#if SDL_THREAD_OS2
+    SDL_OS2TLSAlloc(); /* thread/os2/SDL_systls.c */
+#endif
+
+#if SDL_VIDEO_DRIVER_WINDOWS
+    if (flags & (SDL_INIT_HAPTIC | SDL_INIT_JOYSTICK)) {
+        if (SDL_HelperWindowCreate() < 0) {
+            goto quit_and_error;
+        }
+    }
+#endif
+
+#if !SDL_TIMERS_DISABLED
+    SDL_TicksInit();
+#endif
+
+    /* Initialize the event subsystem */
+    if (flags & SDL_INIT_EVENTS) {
+#if !SDL_EVENTS_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_EVENTS)) {
+            if (SDL_EventsInit() < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_EVENTS);
+        flags_initialized |= SDL_INIT_EVENTS;
+#else
+        SDL_SetError("SDL not built with events support");
+        goto quit_and_error;
+#endif
+    }
+
+    /* Initialize the timer subsystem */
+    if (flags & SDL_INIT_TIMER) {
+#if !SDL_TIMERS_DISABLED && !SDL_TIMER_DUMMY
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_TIMER)) {
+            if (SDL_TimerInit() < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_TIMER);
+        flags_initialized |= SDL_INIT_TIMER;
+#else
+        SDL_SetError("SDL not built with timer support");
+        goto quit_and_error;
+#endif
+    }
+
+    /* Initialize the video subsystem */
+    if (flags & SDL_INIT_VIDEO) {
+#if !SDL_VIDEO_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_VIDEO)) {
+            if (SDL_VideoInit(NULL) < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_VIDEO);
+        flags_initialized |= SDL_INIT_VIDEO;
+#else
+        SDL_SetError("SDL not built with video support");
+        goto quit_and_error;
+#endif
+    }
+
+    /* Initialize the audio subsystem */
+    if (flags & SDL_INIT_AUDIO) {
+#if !SDL_AUDIO_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_AUDIO)) {
+            if (SDL_AudioInit(NULL) < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_AUDIO);
+        flags_initialized |= SDL_INIT_AUDIO;
+#else
+        SDL_SetError("SDL not built with audio support");
+        goto quit_and_error;
+#endif
+    }
+
+    /* Initialize the joystick subsystem */
+    if (flags & SDL_INIT_JOYSTICK) {
+#if !SDL_JOYSTICK_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_JOYSTICK)) {
+            if (SDL_JoystickInit() < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_JOYSTICK);
+        flags_initialized |= SDL_INIT_JOYSTICK;
+#else
+        SDL_SetError("SDL not built with joystick support");
+        goto quit_and_error;
+#endif
+    }
+
+    if (flags & SDL_INIT_GAMECONTROLLER) {
+#if !SDL_JOYSTICK_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_GAMECONTROLLER)) {
+            if (SDL_GameControllerInit() < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_GAMECONTROLLER);
+        flags_initialized |= SDL_INIT_GAMECONTROLLER;
+#else
+        SDL_SetError("SDL not built with joystick support");
+        goto quit_and_error;
+#endif
+    }
+
+    /* Initialize the haptic subsystem */
+    if (flags & SDL_INIT_HAPTIC) {
+#if !SDL_HAPTIC_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_HAPTIC)) {
+            if (SDL_HapticInit() < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_HAPTIC);
+        flags_initialized |= SDL_INIT_HAPTIC;
+#else
+        SDL_SetError("SDL not built with haptic (force feedback) support");
+        goto quit_and_error;
+#endif
+    }
+
+    /* Initialize the sensor subsystem */
+    if (flags & SDL_INIT_SENSOR) {
+#if !SDL_SENSOR_DISABLED
+        if (SDL_PrivateShouldInitSubsystem(SDL_INIT_SENSOR)) {
+            if (SDL_SensorInit() < 0) {
+                goto quit_and_error;
+            }
+        }
+        SDL_PrivateSubsystemRefCountIncr(SDL_INIT_SENSOR);
+        flags_initialized |= SDL_INIT_SENSOR;
+#else
+        SDL_SetError("SDL not built with sensor support");
+        goto quit_and_error;
+#endif
+    }
+
+    (void)flags_initialized; /* make static analysis happy, since this only gets used in error cases. */
+
+    return 0;
+
+quit_and_error:
+    SDL_QuitSubSystem(flags_initialized);
+    return -1;
+}
+
+int SDL_Init(Uint32 flags)
+{
+    return SDL_InitSubSystem(flags);
+}
+
+void SDL_QuitSubSystem(Uint32 flags)
+{
+#if defined(__OS2__)
+#if SDL_THREAD_OS2
+    SDL_OS2TLSFree(); /* thread/os2/SDL_systls.c */
+#endif
+    SDL_OS2Quit();
+#endif
+
+    /* Shut down requested initialized subsystems */
+#if !SDL_SENSOR_DISABLED
+    if (flags & SDL_INIT_SENSOR) {
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_SENSOR)) {
+            SDL_SensorQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_SENSOR);
+    }
+#endif
+
+#if !SDL_JOYSTICK_DISABLED
+    if (flags & SDL_INIT_GAMECONTROLLER) {
+        /* game controller implies joystick */
+        flags |= SDL_INIT_JOYSTICK;
+
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_GAMECONTROLLER)) {
+            SDL_GameControllerQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_GAMECONTROLLER);
+    }
+
+    if (flags & SDL_INIT_JOYSTICK) {
+        /* joystick implies events */
+        flags |= SDL_INIT_EVENTS;
+
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_JOYSTICK)) {
+            SDL_JoystickQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_JOYSTICK);
+    }
+#endif
+
+#if !SDL_HAPTIC_DISABLED
+    if (flags & SDL_INIT_HAPTIC) {
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_HAPTIC)) {
+            SDL_HapticQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_HAPTIC);
+    }
+#endif
+
+#if !SDL_AUDIO_DISABLED
+    if (flags & SDL_INIT_AUDIO) {
+        /* audio implies events */
+        flags |= SDL_INIT_EVENTS;
+
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_AUDIO)) {
+            SDL_AudioQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_AUDIO);
+    }
+#endif
+
+#if !SDL_VIDEO_DISABLED
+    if (flags & SDL_INIT_VIDEO) {
+        /* video implies events */
+        flags |= SDL_INIT_EVENTS;
+
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_VIDEO)) {
+            SDL_VideoQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_VIDEO);
+    }
+#endif
+
+#if !SDL_TIMERS_DISABLED && !SDL_TIMER_DUMMY
+    if (flags & SDL_INIT_TIMER) {
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_TIMER)) {
+            SDL_TimerQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_TIMER);
+    }
+#endif
+
+#if !SDL_EVENTS_DISABLED
+    if (flags & SDL_INIT_EVENTS) {
+        if (SDL_PrivateShouldQuitSubsystem(SDL_INIT_EVENTS)) {
+            SDL_EventsQuit();
+        }
+        SDL_PrivateSubsystemRefCountDecr(SDL_INIT_EVENTS);
+    }
+#endif
+}
+
+Uint32 SDL_WasInit(Uint32 flags)
+{
+    int i;
+    int num_subsystems = SDL_arraysize(SDL_SubsystemRefCount);
+    Uint32 initialized = 0;
+
+    /* Fast path for checking one flag */
+    if (SDL_HasExactlyOneBitSet32(flags)) {
+        int subsystem_index = SDL_MostSignificantBitIndex32(flags);
+        return SDL_SubsystemRefCount[subsystem_index] ? flags : 0;
+    }
+
+    if (!flags) {
+        flags = SDL_INIT_EVERYTHING;
+    }
+
+    num_subsystems = SDL_min(num_subsystems, SDL_MostSignificantBitIndex32(flags) + 1);
+
+    /* Iterate over each bit in flags, and check the matching subsystem. */
+    for (i = 0; i < num_subsystems; ++i) {
+        if ((flags & 1) && SDL_SubsystemRefCount[i] > 0) {
+            initialized |= (1 << i);
+        }
+
+        flags >>= 1;
+    }
+
+    return initialized;
+}
+
+void SDL_Quit(void)
+{
+    SDL_bInMainQuit = SDL_TRUE;
+
+    /* Quit all subsystems */
+#if SDL_VIDEO_DRIVER_WINDOWS
+    SDL_HelperWindowDestroy();
+#endif
+    SDL_QuitSubSystem(SDL_INIT_EVERYTHING);
+
+#if !SDL_TIMERS_DISABLED
+    SDL_TicksQuit();
+#endif
+
+    SDL_ClearHints();
+    SDL_AssertionsQuit();
+
+#if SDL_USE_LIBDBUS
+    SDL_DBus_Quit();
+#endif
+
+    SDL_LogQuit();
+
+    /* Now that every subsystem has been quit, we reset the subsystem refcount
+     * and the list of initialized subsystems.
+     */
+    SDL_memset(SDL_SubsystemRefCount, 0x0, sizeof(SDL_SubsystemRefCount));
+
+    SDL_TLSCleanup();
+
+    SDL_bInMainQuit = SDL_FALSE;
+}
+
+/* Get the library version number */
+void SDL_GetVersion(SDL_version *ver)
+{
+    static SDL_bool check_hint = SDL_TRUE;
+    static SDL_bool legacy_version = SDL_FALSE;
+
+    if (ver == NULL) {
+        return;
+    }
+
+    SDL_VERSION(ver);
+
+    if (check_hint) {
+        check_hint = SDL_FALSE;
+        legacy_version = SDL_GetHintBoolean("SDL_LEGACY_VERSION", SDL_FALSE);
+    }
+
+    if (legacy_version) {
+        /* Prior to SDL 2.24.0, the patch version was incremented with every release */
+        ver->patch = ver->minor;
+        ver->minor = 0;
+    }
+}
+
+/* Get the library source revision */
+const char *SDL_GetRevision(void)
+{
+    return SDL_REVISION;
+}
+
+/* Get the library source revision number */
+int SDL_GetRevisionNumber(void)
+{
+    return 0;  /* doesn't make sense without Mercurial. */
+}
+
+/* Get the name of the platform */
+const char *SDL_GetPlatform(void)
+{
+#if __AIX__
+    return "AIX";
+#elif __ANDROID__
+    return "Android";
+#elif __BSDI__
+    return "BSDI";
+#elif __DREAMCAST__
+    return "Dreamcast";
+#elif __EMSCRIPTEN__
+    return "Emscripten";
+#elif __FREEBSD__
+    return "FreeBSD";
+#elif __HAIKU__
+    return "Haiku";
+#elif __HPUX__
+    return "HP-UX";
+#elif __IRIX__
+    return "Irix";
+#elif __LINUX__
+    return "Linux";
+#elif __MINT__
+    return "Atari MiNT";
+#elif __MACOS__
+    return "MacOS Classic";
+#elif __MACOSX__
+    return "Mac OS X";
+#elif __NACL__
+    return "NaCl";
+#elif __NETBSD__
+    return "NetBSD";
+#elif __OPENBSD__
+    return "OpenBSD";
+#elif __OS2__
+    return "OS/2";
+#elif __OSF__
+    return "OSF/1";
+#elif __QNXNTO__
+    return "QNX Neutrino";
+#elif __RISCOS__
+    return "RISC OS";
+#elif __SOLARIS__
+    return "Solaris";
+#elif __WIN32__
+    return "Windows";
+#elif __WINRT__
+    return "WinRT";
+#elif __WINGDK__
+    return "WinGDK";
+#elif __XBOXONE__
+    return "Xbox One";
+#elif __XBOXSERIES__
+    return "Xbox Series X|S";
+#elif __TVOS__
+    return "tvOS";
+#elif __IPHONEOS__
+    return "iOS";
+#elif __PS2__
+    return "PlayStation 2";
+#elif __PSP__
+    return "PlayStation Portable";
+#elif __VITA__
+    return "PlayStation Vita";
+#elif __NGAGE__
+    return "Nokia N-Gage";
+#elif __3DS__
+    return "Nintendo 3DS";
+#else
+    return "Unknown (see SDL_platform.h)";
+#endif
+}
+
+SDL_bool SDL_IsTablet(void)
+{
+#if __ANDROID__
+    extern SDL_bool SDL_IsAndroidTablet(void);
+    return SDL_IsAndroidTablet();
+#elif __IPHONEOS__
+    extern SDL_bool SDL_IsIPad(void);
+    return SDL_IsIPad();
+#else
+    return SDL_FALSE;
+#endif
+}
+
+#if defined(__WIN32__)
+
+#if (!defined(HAVE_LIBC) || defined(__WATCOMC__)) && !defined(SDL_STATIC_LIB)
+/* FIXME: Still need to include DllMain() on Watcom C ? */
+
+BOOL APIENTRY MINGW32_FORCEALIGN _DllMainCRTStartup(HANDLE hModule, DWORD ul_reason_for_call, LPVOID lpReserved)
+{
+    switch (ul_reason_for_call) {
+    case DLL_PROCESS_ATTACH:
+    case DLL_THREAD_ATTACH:
+    case DLL_THREAD_DETACH:
+    case DLL_PROCESS_DETACH:
+        break;
+    }
+    return TRUE;
+}
+#endif /* Building DLL */
+
+#endif /* defined(__WIN32__) || defined(__GDK__) */
+
+/* vi: set sts=4 ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_assert.c

@@ -0,0 +1,463 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+#if defined(__WIN32__) || defined(__GDK__)
+#include "core/windows/SDL_windows.h"
+#endif
+
+#include "SDL.h"
+#include "SDL_atomic.h"
+#include "SDL_messagebox.h"
+#include "SDL_video.h"
+#include "SDL_assert.h"
+#include "SDL_assert_c.h"
+#include "video/SDL_sysvideo.h"
+
+#if defined(__WIN32__) || defined(__GDK__)
+#ifndef WS_OVERLAPPEDWINDOW
+#define WS_OVERLAPPEDWINDOW 0
+#endif
+#else  /* fprintf, etc. */
+#include <stdio.h>
+#include <stdlib.h>
+#endif
+
+#if defined(__EMSCRIPTEN__)
+    #include <emscripten.h>
+    /* older Emscriptens don't have this, but we need to for wasm64 compatibility. */
+    #ifndef MAIN_THREAD_EM_ASM_PTR
+        #ifdef __wasm64__
+            #error You need to upgrade your Emscripten compiler to support wasm64
+        #else
+            #define MAIN_THREAD_EM_ASM_PTR MAIN_THREAD_EM_ASM_INT
+        #endif
+    #endif
+#endif
+
+/* The size of the stack buffer to use for rendering assert messages. */
+#define SDL_MAX_ASSERT_MESSAGE_STACK 256
+
+static SDL_assert_state SDLCALL SDL_PromptAssertion(const SDL_assert_data *data, void *userdata);
+
+/*
+ * We keep all triggered assertions in a singly-linked list so we can
+ *  generate a report later.
+ */
+static SDL_assert_data *triggered_assertions = NULL;
+
+#ifndef SDL_THREADS_DISABLED
+static SDL_mutex *assertion_mutex = NULL;
+#endif
+
+static SDL_AssertionHandler assertion_handler = SDL_PromptAssertion;
+static void *assertion_userdata = NULL;
+
+#ifdef __GNUC__
+static void debug_print(const char *fmt, ...) __attribute__((format(printf, 1, 2)));
+#endif
+
+static void debug_print(const char *fmt, ...)
+{
+    va_list ap;
+    va_start(ap, fmt);
+    SDL_LogMessageV(SDL_LOG_CATEGORY_ASSERT, SDL_LOG_PRIORITY_WARN, fmt, ap);
+    va_end(ap);
+}
+
+static void SDL_AddAssertionToReport(SDL_assert_data *data)
+{
+    /* (data) is always a static struct defined with the assert macros, so
+       we don't have to worry about copying or allocating them. */
+    data->trigger_count++;
+    if (data->trigger_count == 1) { /* not yet added? */
+        data->next = triggered_assertions;
+        triggered_assertions = data;
+    }
+}
+
+#if defined(__WIN32__) || defined(__GDK__)
+#define ENDLINE "\r\n"
+#else
+#define ENDLINE "\n"
+#endif
+
+static int SDL_RenderAssertMessage(char *buf, size_t buf_len, const SDL_assert_data *data)
+{
+    return SDL_snprintf(buf, buf_len,
+                        "Assertion failure at %s (%s:%d), triggered %u %s:" ENDLINE "  '%s'",
+                        data->function, data->filename, data->linenum,
+                        data->trigger_count, (data->trigger_count == 1) ? "time" : "times",
+                        data->condition);
+}
+
+static void SDL_GenerateAssertionReport(void)
+{
+    const SDL_assert_data *item = triggered_assertions;
+
+    /* only do this if the app hasn't assigned an assertion handler. */
+    if ((item != NULL) && (assertion_handler != SDL_PromptAssertion)) {
+        debug_print("\n\nSDL assertion report.\n");
+        debug_print("All SDL assertions between last init/quit:\n\n");
+
+        while (item != NULL) {
+            debug_print(
+                "'%s'\n"
+                "    * %s (%s:%d)\n"
+                "    * triggered %u time%s.\n"
+                "    * always ignore: %s.\n",
+                item->condition, item->function, item->filename,
+                item->linenum, item->trigger_count,
+                (item->trigger_count == 1) ? "" : "s",
+                item->always_ignore ? "yes" : "no");
+            item = item->next;
+        }
+        debug_print("\n");
+
+        SDL_ResetAssertionReport();
+    }
+}
+
+/* This is not declared in any header, although it is shared between some
+    parts of SDL, because we don't want anything calling it without an
+    extremely good reason. */
+#if defined(__WATCOMC__)
+extern void SDL_ExitProcess(int exitcode);
+#pragma aux SDL_ExitProcess aborts;
+#endif
+extern SDL_NORETURN void SDL_ExitProcess(int exitcode);
+
+#if defined(__WATCOMC__)
+static void SDL_AbortAssertion(void);
+#pragma aux SDL_AbortAssertion aborts;
+#endif
+static SDL_NORETURN void SDL_AbortAssertion(void)
+{
+    SDL_Quit();
+    SDL_ExitProcess(42);
+}
+
+static SDL_assert_state SDLCALL SDL_PromptAssertion(const SDL_assert_data *data, void *userdata)
+{
+    const char *envr;
+    SDL_assert_state state = SDL_ASSERTION_ABORT;
+    SDL_Window *window;
+    SDL_MessageBoxData messagebox;
+    SDL_MessageBoxButtonData buttons[] = {
+        { 0, SDL_ASSERTION_RETRY, "Retry" },
+        { 0, SDL_ASSERTION_BREAK, "Break" },
+        { 0, SDL_ASSERTION_ABORT, "Abort" },
+        { SDL_MESSAGEBOX_BUTTON_ESCAPEKEY_DEFAULT,
+          SDL_ASSERTION_IGNORE, "Ignore" },
+        { SDL_MESSAGEBOX_BUTTON_RETURNKEY_DEFAULT,
+          SDL_ASSERTION_ALWAYS_IGNORE, "Always Ignore" }
+    };
+    int selected;
+
+    char stack_buf[SDL_MAX_ASSERT_MESSAGE_STACK];
+    char *message = stack_buf;
+    size_t buf_len = sizeof(stack_buf);
+    int len;
+
+    (void)userdata; /* unused in default handler. */
+
+    /* Assume the output will fit... */
+    len = SDL_RenderAssertMessage(message, buf_len, data);
+
+    /* .. and if it didn't, try to allocate as much room as we actually need. */
+    if (len >= (int)buf_len) {
+        if (SDL_size_add_overflow(len, 1, &buf_len) == 0) {
+            message = (char *)SDL_malloc(buf_len);
+            if (message) {
+                len = SDL_RenderAssertMessage(message, buf_len, data);
+            } else {
+                message = stack_buf;
+            }
+        }
+    }
+
+    /* Something went very wrong */
+    if (len < 0) {
+        if (message != stack_buf) {
+            SDL_free(message);
+        }
+        return SDL_ASSERTION_ABORT;
+    }
+
+    debug_print("\n\n%s\n\n", message);
+
+    /* let env. variable override, so unit tests won't block in a GUI. */
+    envr = SDL_getenv("SDL_ASSERT");
+    if (envr != NULL) {
+        if (message != stack_buf) {
+            SDL_free(message);
+        }
+
+        if (SDL_strcmp(envr, "abort") == 0) {
+            return SDL_ASSERTION_ABORT;
+        } else if (SDL_strcmp(envr, "break") == 0) {
+            return SDL_ASSERTION_BREAK;
+        } else if (SDL_strcmp(envr, "retry") == 0) {
+            return SDL_ASSERTION_RETRY;
+        } else if (SDL_strcmp(envr, "ignore") == 0) {
+            return SDL_ASSERTION_IGNORE;
+        } else if (SDL_strcmp(envr, "always_ignore") == 0) {
+            return SDL_ASSERTION_ALWAYS_IGNORE;
+        } else {
+            return SDL_ASSERTION_ABORT; /* oh well. */
+        }
+    }
+
+    /* Leave fullscreen mode, if possible (scary!) */
+    window = SDL_GetFocusWindow();
+    if (window) {
+        if (SDL_GetWindowFlags(window) & SDL_WINDOW_FULLSCREEN) {
+            SDL_MinimizeWindow(window);
+        } else {
+            /* !!! FIXME: ungrab the input if we're not fullscreen? */
+            /* No need to mess with the window */
+            window = NULL;
+        }
+    }
+
+    /* Show a messagebox if we can, otherwise fall back to stdio */
+    SDL_zero(messagebox);
+    messagebox.flags = SDL_MESSAGEBOX_WARNING;
+    messagebox.window = window;
+    messagebox.title = "Assertion Failed";
+    messagebox.message = message;
+    messagebox.numbuttons = SDL_arraysize(buttons);
+    messagebox.buttons = buttons;
+
+    if (SDL_ShowMessageBox(&messagebox, &selected) == 0) {
+        if (selected == -1) {
+            state = SDL_ASSERTION_IGNORE;
+        } else {
+            state = (SDL_assert_state)selected;
+        }
+    } else {
+#if defined(__EMSCRIPTEN__)
+        /* This is nasty, but we can't block on a custom UI. */
+        for (;;) {
+            SDL_bool okay = SDL_TRUE;
+            /* *INDENT-OFF* */ /* clang-format off */
+            char *buf = (char *) MAIN_THREAD_EM_ASM_PTR({
+                var str =
+                    UTF8ToString($0) + '\n\n' +
+                    'Abort/Retry/Ignore/AlwaysIgnore? [ariA] :';
+                var reply = window.prompt(str, "i");
+                if (reply === null) {
+                    reply = "i";
+                }
+                return allocate(intArrayFromString(reply), 'i8', ALLOC_NORMAL);
+            }, message);
+            /* *INDENT-ON* */ /* clang-format on */
+
+            if (SDL_strcmp(buf, "a") == 0) {
+                state = SDL_ASSERTION_ABORT;
+#if 0 /* (currently) no break functionality on Emscripten */
+            } else if (SDL_strcmp(buf, "b") == 0) {
+                state = SDL_ASSERTION_BREAK;
+#endif
+            } else if (SDL_strcmp(buf, "r") == 0) {
+                state = SDL_ASSERTION_RETRY;
+            } else if (SDL_strcmp(buf, "i") == 0) {
+                state = SDL_ASSERTION_IGNORE;
+            } else if (SDL_strcmp(buf, "A") == 0) {
+                state = SDL_ASSERTION_ALWAYS_IGNORE;
+            } else {
+                okay = SDL_FALSE;
+            }
+            free(buf);
+
+            if (okay) {
+                break;
+            }
+        }
+#elif defined(HAVE_STDIO_H)
+        /* this is a little hacky. */
+        for (;;) {
+            char buf[32];
+            (void)fprintf(stderr, "Abort/Break/Retry/Ignore/AlwaysIgnore? [abriA] : ");
+            (void)fflush(stderr);
+            if (fgets(buf, sizeof(buf), stdin) == NULL) {
+                break;
+            }
+
+            if (SDL_strncmp(buf, "a", 1) == 0) {
+                state = SDL_ASSERTION_ABORT;
+                break;
+            } else if (SDL_strncmp(buf, "b", 1) == 0) {
+                state = SDL_ASSERTION_BREAK;
+                break;
+            } else if (SDL_strncmp(buf, "r", 1) == 0) {
+                state = SDL_ASSERTION_RETRY;
+                break;
+            } else if (SDL_strncmp(buf, "i", 1) == 0) {
+                state = SDL_ASSERTION_IGNORE;
+                break;
+            } else if (SDL_strncmp(buf, "A", 1) == 0) {
+                state = SDL_ASSERTION_ALWAYS_IGNORE;
+                break;
+            }
+        }
+#endif /* HAVE_STDIO_H */
+    }
+
+    /* Re-enter fullscreen mode */
+    if (window) {
+        SDL_RestoreWindow(window);
+    }
+
+    if (message != stack_buf) {
+        SDL_free(message);
+    }
+
+    return state;
+}
+
+SDL_assert_state SDL_ReportAssertion(SDL_assert_data *data, const char *func, const char *file, int line)
+{
+    SDL_assert_state state = SDL_ASSERTION_IGNORE;
+    static int assertion_running = 0;
+
+#ifndef SDL_THREADS_DISABLED
+    static SDL_SpinLock spinlock = 0;
+    SDL_AtomicLock(&spinlock);
+    if (assertion_mutex == NULL) { /* never called SDL_Init()? */
+        assertion_mutex = SDL_CreateMutex();
+        if (assertion_mutex == NULL) {
+            SDL_AtomicUnlock(&spinlock);
+            return SDL_ASSERTION_IGNORE; /* oh well, I guess. */
+        }
+    }
+    SDL_AtomicUnlock(&spinlock);
+
+    SDL_LockMutex(assertion_mutex);
+#endif /* !SDL_THREADS_DISABLED */
+
+    /* doing this because Visual C is upset over assigning in the macro. */
+    if (data->trigger_count == 0) {
+        data->function = func;
+        data->filename = file;
+        data->linenum = line;
+    }
+
+    SDL_AddAssertionToReport(data);
+
+    assertion_running++;
+    if (assertion_running > 1) { /* assert during assert! Abort. */
+        if (assertion_running == 2) {
+            SDL_AbortAssertion();
+        } else if (assertion_running == 3) { /* Abort asserted! */
+            SDL_ExitProcess(42);
+        } else {
+            while (1) { /* do nothing but spin; what else can you do?! */
+            }
+        }
+    }
+
+    if (!data->always_ignore) {
+        state = assertion_handler(data, assertion_userdata);
+    }
+
+    switch (state) {
+    case SDL_ASSERTION_ALWAYS_IGNORE:
+        state = SDL_ASSERTION_IGNORE;
+        data->always_ignore = 1;
+        break;
+
+    case SDL_ASSERTION_IGNORE:
+    case SDL_ASSERTION_RETRY:
+    case SDL_ASSERTION_BREAK:
+        break; /* macro handles these. */
+
+    case SDL_ASSERTION_ABORT:
+        SDL_AbortAssertion();
+        /*break;  ...shouldn't return, but oh well. */
+    }
+
+    assertion_running--;
+
+#ifndef SDL_THREADS_DISABLED
+    SDL_UnlockMutex(assertion_mutex);
+#endif
+
+    return state;
+}
+
+void SDL_AssertionsQuit(void)
+{
+#if SDL_ASSERT_LEVEL > 0
+    SDL_GenerateAssertionReport();
+#ifndef SDL_THREADS_DISABLED
+    if (assertion_mutex != NULL) {
+        SDL_DestroyMutex(assertion_mutex);
+        assertion_mutex = NULL;
+    }
+#endif
+#endif /* SDL_ASSERT_LEVEL > 0 */
+}
+
+void SDL_SetAssertionHandler(SDL_AssertionHandler handler, void *userdata)
+{
+    if (handler != NULL) {
+        assertion_handler = handler;
+        assertion_userdata = userdata;
+    } else {
+        assertion_handler = SDL_PromptAssertion;
+        assertion_userdata = NULL;
+    }
+}
+
+const SDL_assert_data *SDL_GetAssertionReport(void)
+{
+    return triggered_assertions;
+}
+
+void SDL_ResetAssertionReport(void)
+{
+    SDL_assert_data *next = NULL;
+    SDL_assert_data *item;
+    for (item = triggered_assertions; item != NULL; item = next) {
+        next = (SDL_assert_data *)item->next;
+        item->always_ignore = SDL_FALSE;
+        item->trigger_count = 0;
+        item->next = NULL;
+    }
+
+    triggered_assertions = NULL;
+}
+
+SDL_AssertionHandler SDL_GetDefaultAssertionHandler(void)
+{
+    return SDL_PromptAssertion;
+}
+
+SDL_AssertionHandler SDL_GetAssertionHandler(void **userdata)
+{
+    if (userdata != NULL) {
+        *userdata = assertion_userdata;
+    }
+    return assertion_handler;
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_assert_c.h

@@ -0,0 +1,29 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_assert_c_h_
+#define SDL_assert_c_h_
+
+extern void SDL_AssertionsQuit(void);
+
+#endif /* SDL_assert_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_dataqueue.c

@@ -0,0 +1,323 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+#include "SDL.h"
+#include "./SDL_dataqueue.h"
+
+typedef struct SDL_DataQueuePacket
+{
+    size_t datalen;                        /* bytes currently in use in this packet. */
+    size_t startpos;                       /* bytes currently consumed in this packet. */
+    struct SDL_DataQueuePacket *next;      /* next item in linked list. */
+    Uint8 data[SDL_VARIABLE_LENGTH_ARRAY]; /* packet data */
+} SDL_DataQueuePacket;
+
+struct SDL_DataQueue
+{
+    SDL_mutex *lock;
+    SDL_DataQueuePacket *head; /* device fed from here. */
+    SDL_DataQueuePacket *tail; /* queue fills to here. */
+    SDL_DataQueuePacket *pool; /* these are unused packets. */
+    size_t packet_size;        /* size of new packets */
+    size_t queued_bytes;       /* number of bytes of data in the queue. */
+};
+
+static void SDL_FreeDataQueueList(SDL_DataQueuePacket *packet)
+{
+    while (packet) {
+        SDL_DataQueuePacket *next = packet->next;
+        SDL_free(packet);
+        packet = next;
+    }
+}
+
+SDL_DataQueue *SDL_NewDataQueue(const size_t _packetlen, const size_t initialslack)
+{
+    SDL_DataQueue *queue = (SDL_DataQueue *)SDL_calloc(1, sizeof(SDL_DataQueue));
+
+    if (queue == NULL) {
+        SDL_OutOfMemory();
+    } else {
+        const size_t packetlen = _packetlen ? _packetlen : 1024;
+        const size_t wantpackets = (initialslack + (packetlen - 1)) / packetlen;
+        size_t i;
+
+        queue->packet_size = packetlen;
+
+        queue->lock = SDL_CreateMutex();
+        if (!queue->lock) {
+            SDL_free(queue);
+            return NULL;
+        }
+
+        for (i = 0; i < wantpackets; i++) {
+            SDL_DataQueuePacket *packet = (SDL_DataQueuePacket *)SDL_malloc(sizeof(SDL_DataQueuePacket) + packetlen);
+            if (packet) { /* don't care if this fails, we'll deal later. */
+                packet->datalen = 0;
+                packet->startpos = 0;
+                packet->next = queue->pool;
+                queue->pool = packet;
+            }
+        }
+    }
+
+    return queue;
+}
+
+void SDL_FreeDataQueue(SDL_DataQueue *queue)
+{
+    if (queue) {
+        SDL_FreeDataQueueList(queue->head);
+        SDL_FreeDataQueueList(queue->pool);
+        SDL_DestroyMutex(queue->lock);
+        SDL_free(queue);
+    }
+}
+
+void SDL_ClearDataQueue(SDL_DataQueue *queue, const size_t slack)
+{
+    const size_t packet_size = queue ? queue->packet_size : 1;
+    const size_t slackpackets = (slack + (packet_size - 1)) / packet_size;
+    SDL_DataQueuePacket *packet;
+    SDL_DataQueuePacket *prev = NULL;
+    size_t i;
+
+    if (queue == NULL) {
+        return;
+    }
+
+    SDL_LockMutex(queue->lock);
+
+    packet = queue->head;
+
+    /* merge the available pool and the current queue into one list. */
+    if (packet) {
+        queue->tail->next = queue->pool;
+    } else {
+        packet = queue->pool;
+    }
+
+    /* Remove the queued packets from the device. */
+    queue->tail = NULL;
+    queue->head = NULL;
+    queue->queued_bytes = 0;
+    queue->pool = packet;
+
+    /* Optionally keep some slack in the pool to reduce memory allocation pressure. */
+    for (i = 0; packet && (i < slackpackets); i++) {
+        prev = packet;
+        packet = packet->next;
+    }
+
+    if (prev) {
+        prev->next = NULL;
+    } else {
+        queue->pool = NULL;
+    }
+
+    SDL_UnlockMutex(queue->lock);
+
+    SDL_FreeDataQueueList(packet); /* free extra packets */
+}
+
+/* You must hold queue->lock before calling this! */
+static SDL_DataQueuePacket *AllocateDataQueuePacket(SDL_DataQueue *queue)
+{
+    SDL_DataQueuePacket *packet;
+
+    SDL_assert(queue != NULL);
+
+    packet = queue->pool;
+    if (packet != NULL) {
+        /* we have one available in the pool. */
+        queue->pool = packet->next;
+    } else {
+        /* Have to allocate a new one! */
+        packet = (SDL_DataQueuePacket *)SDL_malloc(sizeof(SDL_DataQueuePacket) + queue->packet_size);
+        if (packet == NULL) {
+            return NULL;
+        }
+    }
+
+    packet->datalen = 0;
+    packet->startpos = 0;
+    packet->next = NULL;
+
+    SDL_assert((queue->head != NULL) == (queue->queued_bytes != 0));
+    if (queue->tail == NULL) {
+        queue->head = packet;
+    } else {
+        queue->tail->next = packet;
+    }
+    queue->tail = packet;
+    return packet;
+}
+
+int SDL_WriteToDataQueue(SDL_DataQueue *queue, const void *_data, const size_t _len)
+{
+    size_t len = _len;
+    const Uint8 *data = (const Uint8 *)_data;
+    const size_t packet_size = queue ? queue->packet_size : 0;
+    SDL_DataQueuePacket *orighead;
+    SDL_DataQueuePacket *origtail;
+    size_t origlen;
+    size_t datalen;
+
+    if (queue == NULL) {
+        return SDL_InvalidParamError("queue");
+    }
+
+    SDL_LockMutex(queue->lock);
+
+    orighead = queue->head;
+    origtail = queue->tail;
+    origlen = origtail ? origtail->datalen : 0;
+
+    while (len > 0) {
+        SDL_DataQueuePacket *packet = queue->tail;
+        SDL_assert(packet == NULL || (packet->datalen <= packet_size));
+        if (packet == NULL || (packet->datalen >= packet_size)) {
+            /* tail packet missing or completely full; we need a new packet. */
+            packet = AllocateDataQueuePacket(queue);
+            if (packet == NULL) {
+                /* uhoh, reset so we've queued nothing new, free what we can. */
+                if (origtail == NULL) {
+                    packet = queue->head; /* whole queue. */
+                } else {
+                    packet = origtail->next; /* what we added to existing queue. */
+                    origtail->next = NULL;
+                    origtail->datalen = origlen;
+                }
+                queue->head = orighead;
+                queue->tail = origtail;
+                queue->pool = NULL;
+
+                SDL_UnlockMutex(queue->lock);
+                SDL_FreeDataQueueList(packet); /* give back what we can. */
+                return SDL_OutOfMemory();
+            }
+        }
+
+        datalen = SDL_min(len, packet_size - packet->datalen);
+        SDL_memcpy(packet->data + packet->datalen, data, datalen);
+        data += datalen;
+        len -= datalen;
+        packet->datalen += datalen;
+        queue->queued_bytes += datalen;
+    }
+
+    SDL_UnlockMutex(queue->lock);
+
+    return 0;
+}
+
+size_t
+SDL_PeekIntoDataQueue(SDL_DataQueue *queue, void *_buf, const size_t _len)
+{
+    size_t len = _len;
+    Uint8 *buf = (Uint8 *)_buf;
+    Uint8 *ptr = buf;
+    SDL_DataQueuePacket *packet;
+
+    if (queue == NULL) {
+        return 0;
+    }
+
+    SDL_LockMutex(queue->lock);
+
+    for (packet = queue->head; len && packet; packet = packet->next) {
+        const size_t avail = packet->datalen - packet->startpos;
+        const size_t cpy = SDL_min(len, avail);
+        SDL_assert(queue->queued_bytes >= avail);
+
+        SDL_memcpy(ptr, packet->data + packet->startpos, cpy);
+        ptr += cpy;
+        len -= cpy;
+    }
+
+    SDL_UnlockMutex(queue->lock);
+
+    return (size_t)(ptr - buf);
+}
+
+size_t
+SDL_ReadFromDataQueue(SDL_DataQueue *queue, void *_buf, const size_t _len)
+{
+    size_t len = _len;
+    Uint8 *buf = (Uint8 *)_buf;
+    Uint8 *ptr = buf;
+    SDL_DataQueuePacket *packet;
+
+    if (queue == NULL) {
+        return 0;
+    }
+
+    SDL_LockMutex(queue->lock);
+
+    while ((len > 0) && ((packet = queue->head) != NULL)) {
+        const size_t avail = packet->datalen - packet->startpos;
+        const size_t cpy = SDL_min(len, avail);
+        SDL_assert(queue->queued_bytes >= avail);
+
+        SDL_memcpy(ptr, packet->data + packet->startpos, cpy);
+        packet->startpos += cpy;
+        ptr += cpy;
+        queue->queued_bytes -= cpy;
+        len -= cpy;
+
+        if (packet->startpos == packet->datalen) { /* packet is done, put it in the pool. */
+            queue->head = packet->next;
+            SDL_assert((packet->next != NULL) || (packet == queue->tail));
+            packet->next = queue->pool;
+            queue->pool = packet;
+        }
+    }
+
+    SDL_assert((queue->head != NULL) == (queue->queued_bytes != 0));
+
+    if (queue->head == NULL) {
+        queue->tail = NULL; /* in case we drained the queue entirely. */
+    }
+
+    SDL_UnlockMutex(queue->lock);
+
+    return (size_t)(ptr - buf);
+}
+
+size_t
+SDL_CountDataQueue(SDL_DataQueue *queue)
+{
+    size_t retval = 0;
+    if (queue) {
+        SDL_LockMutex(queue->lock);
+        retval = queue->queued_bytes;
+        SDL_UnlockMutex(queue->lock);
+    }
+    return retval;
+}
+
+SDL_mutex *SDL_GetDataQueueMutex(SDL_DataQueue *queue)
+{
+    return queue ? queue->lock : NULL;
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_dataqueue.h

@@ -0,0 +1,40 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#ifndef SDL_dataqueue_h_
+#define SDL_dataqueue_h_
+
+/* this is not (currently) a public API. But maybe it should be! */
+
+struct SDL_DataQueue;
+typedef struct SDL_DataQueue SDL_DataQueue;
+
+SDL_DataQueue *SDL_NewDataQueue(const size_t packetlen, const size_t initialslack);
+void SDL_FreeDataQueue(SDL_DataQueue *queue);
+void SDL_ClearDataQueue(SDL_DataQueue *queue, const size_t slack);
+int SDL_WriteToDataQueue(SDL_DataQueue *queue, const void *data, const size_t len);
+size_t SDL_ReadFromDataQueue(SDL_DataQueue *queue, void *buf, const size_t len);
+size_t SDL_PeekIntoDataQueue(SDL_DataQueue *queue, void *buf, const size_t len);
+size_t SDL_CountDataQueue(SDL_DataQueue *queue);
+SDL_mutex *SDL_GetDataQueueMutex(SDL_DataQueue *queue);  /* don't destroy this, obviously. */
+
+#endif /* SDL_dataqueue_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_error.c

@@ -0,0 +1,123 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+/* Simple error handling in SDL */
+
+#include "SDL_error.h"
+#include "SDL_error_c.h"
+
+int SDL_SetError(SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    /* Ignore call if invalid format pointer was passed */
+    if (fmt != NULL) {
+        va_list ap;
+        int result;
+        SDL_error *error = SDL_GetErrBuf();
+
+        error->error = 1; /* mark error as valid */
+
+        va_start(ap, fmt);
+        result = SDL_vsnprintf(error->str, error->len, fmt, ap);
+        va_end(ap);
+
+        if (result >= 0 && (size_t)result >= error->len && error->realloc_func) {
+            size_t len = (size_t)result + 1;
+            char *str = (char *)error->realloc_func(error->str, len);
+            if (str) {
+                error->str = str;
+                error->len = len;
+                va_start(ap, fmt);
+                (void)SDL_vsnprintf(error->str, error->len, fmt, ap);
+                va_end(ap);
+            }
+        }
+
+        if (SDL_LogGetPriority(SDL_LOG_CATEGORY_ERROR) <= SDL_LOG_PRIORITY_DEBUG) {
+            /* If we are in debug mode, print out the error message */
+            SDL_LogDebug(SDL_LOG_CATEGORY_ERROR, "%s", error->str);
+        }
+    }
+
+    return -1;
+}
+
+/* Available for backwards compatibility */
+const char *SDL_GetError(void)
+{
+    const SDL_error *error = SDL_GetErrBuf();
+    return error->error ? error->str : "";
+}
+
+void SDL_ClearError(void)
+{
+    SDL_GetErrBuf()->error = 0;
+}
+
+/* Very common errors go here */
+int SDL_Error(SDL_errorcode code)
+{
+    switch (code) {
+    case SDL_ENOMEM:
+        return SDL_SetError("Out of memory");
+    case SDL_EFREAD:
+        return SDL_SetError("Error reading from datastream");
+    case SDL_EFWRITE:
+        return SDL_SetError("Error writing to datastream");
+    case SDL_EFSEEK:
+        return SDL_SetError("Error seeking in datastream");
+    case SDL_UNSUPPORTED:
+        return SDL_SetError("That operation is not supported");
+    default:
+        return SDL_SetError("Unknown SDL error");
+    }
+}
+
+#ifdef TEST_ERROR
+int main(int argc, char *argv[])
+{
+    char buffer[BUFSIZ + 1];
+
+    SDL_SetError("Hi there!");
+    printf("Error 1: %s\n", SDL_GetError());
+    SDL_ClearError();
+    SDL_memset(buffer, '1', BUFSIZ);
+    buffer[BUFSIZ] = 0;
+    SDL_SetError("This is the error: %s (%f)", buffer, 1.0);
+    printf("Error 2: %s\n", SDL_GetError());
+    exit(0);
+}
+#endif
+
+char *SDL_GetErrorMsg(char *errstr, int maxlen)
+{
+    const SDL_error *error = SDL_GetErrBuf();
+
+    if (error->error) {
+        SDL_strlcpy(errstr, error->str, maxlen);
+    } else {
+        *errstr = '\0';
+    }
+
+    return errstr;
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_error_c.h

@@ -0,0 +1,44 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+/* This file defines a structure that carries language-independent
+   error messages
+*/
+
+#ifndef SDL_error_c_h_
+#define SDL_error_c_h_
+
+typedef struct SDL_error
+{
+    int error; /* This is a numeric value corresponding to the current error */
+    char *str;
+    size_t len;
+    SDL_realloc_func realloc_func;
+    SDL_free_func free_func;
+} SDL_error;
+
+/* Defined in SDL_thread.c */
+extern SDL_error *SDL_GetErrBuf(void);
+
+#endif /* SDL_error_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_guid.c

@@ -0,0 +1,91 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+
+#include "SDL_guid.h"
+
+/* convert the guid to a printable string */
+void SDL_GUIDToString(SDL_GUID guid, char *pszGUID, int cbGUID)
+{
+    static const char k_rgchHexToASCII[] = "0123456789abcdef";
+    int i;
+
+    if ((pszGUID == NULL) || (cbGUID <= 0)) {
+        return;
+    }
+
+    for (i = 0; i < sizeof(guid.data) && i < (cbGUID - 1) / 2; i++) {
+        /* each input byte writes 2 ascii chars, and might write a null byte. */
+        /* If we don't have room for next input byte, stop */
+        unsigned char c = guid.data[i];
+
+        *pszGUID++ = k_rgchHexToASCII[c >> 4];
+        *pszGUID++ = k_rgchHexToASCII[c & 0x0F];
+    }
+    *pszGUID = '\0';
+}
+
+/*-----------------------------------------------------------------------------
+ * Purpose: Returns the 4 bit nibble for a hex character
+ * Input  : c -
+ * Output : unsigned char
+ *-----------------------------------------------------------------------------*/
+static unsigned char nibble(unsigned char c)
+{
+    if ((c >= '0') && (c <= '9')) {
+        return c - '0';
+    }
+
+    if ((c >= 'A') && (c <= 'F')) {
+        return c - 'A' + 0x0a;
+    }
+
+    if ((c >= 'a') && (c <= 'f')) {
+        return c - 'a' + 0x0a;
+    }
+
+    /* received an invalid character, and no real way to return an error */
+    /* AssertMsg1(false, "Q_nibble invalid hex character '%c' ", c); */
+    return 0;
+}
+
+/* convert the string version of a guid to the struct */
+SDL_GUID SDL_GUIDFromString(const char *pchGUID)
+{
+    SDL_GUID guid;
+    int maxoutputbytes = sizeof(guid);
+    size_t len = SDL_strlen(pchGUID);
+    Uint8 *p;
+    size_t i;
+
+    /* Make sure it's even */
+    len = (len) & ~0x1;
+
+    SDL_memset(&guid, 0x00, sizeof(guid));
+
+    p = (Uint8 *)&guid;
+    for (i = 0; (i < len) && ((p - (Uint8 *)&guid) < maxoutputbytes); i += 2, p++) {
+        *p = (nibble((unsigned char)pchGUID[i]) << 4) | nibble((unsigned char)pchGUID[i + 1]);
+    }
+
+    return guid;
+}

third_party/SDL/src/SDL_hints.c

@@ -0,0 +1,299 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+#include "SDL_hints.h"
+#include "SDL_error.h"
+#include "SDL_hints_c.h"
+
+/* Assuming there aren't many hints set and they aren't being queried in
+   critical performance paths, we'll just use linked lists here.
+ */
+typedef struct SDL_HintWatch
+{
+    SDL_HintCallback callback;
+    void *userdata;
+    struct SDL_HintWatch *next;
+} SDL_HintWatch;
+
+typedef struct SDL_Hint
+{
+    char *name;
+    char *value;
+    SDL_HintPriority priority;
+    SDL_HintWatch *callbacks;
+    struct SDL_Hint *next;
+} SDL_Hint;
+
+static SDL_Hint *SDL_hints;
+
+SDL_bool SDL_SetHintWithPriority(const char *name, const char *value, SDL_HintPriority priority)
+{
+    const char *env;
+    SDL_Hint *hint;
+    SDL_HintWatch *entry;
+
+    if (name == NULL) {
+        return SDL_FALSE;
+    }
+
+    env = SDL_getenv(name);
+    if (env && priority < SDL_HINT_OVERRIDE) {
+        return SDL_FALSE;
+    }
+
+    for (hint = SDL_hints; hint; hint = hint->next) {
+        if (SDL_strcmp(name, hint->name) == 0) {
+            if (priority < hint->priority) {
+                return SDL_FALSE;
+            }
+            if (hint->value != value &&
+                (value == NULL || !hint->value || SDL_strcmp(hint->value, value) != 0)) {
+                for (entry = hint->callbacks; entry;) {
+                    /* Save the next entry in case this one is deleted */
+                    SDL_HintWatch *next = entry->next;
+                    entry->callback(entry->userdata, name, hint->value, value);
+                    entry = next;
+                }
+                SDL_free(hint->value);
+                hint->value = value ? SDL_strdup(value) : NULL;
+            }
+            hint->priority = priority;
+            return SDL_TRUE;
+        }
+    }
+
+    /* Couldn't find the hint, add a new one */
+    hint = (SDL_Hint *)SDL_malloc(sizeof(*hint));
+    if (hint == NULL) {
+        return SDL_FALSE;
+    }
+    hint->name = SDL_strdup(name);
+    hint->value = value ? SDL_strdup(value) : NULL;
+    hint->priority = priority;
+    hint->callbacks = NULL;
+    hint->next = SDL_hints;
+    SDL_hints = hint;
+    return SDL_TRUE;
+}
+
+SDL_bool SDL_ResetHint(const char *name)
+{
+    const char *env;
+    SDL_Hint *hint;
+    SDL_HintWatch *entry;
+
+    if (name == NULL) {
+        return SDL_FALSE;
+    }
+
+    env = SDL_getenv(name);
+    for (hint = SDL_hints; hint; hint = hint->next) {
+        if (SDL_strcmp(name, hint->name) == 0) {
+            if ((env == NULL && hint->value != NULL) ||
+                (env != NULL && hint->value == NULL) ||
+                (env != NULL && SDL_strcmp(env, hint->value) != 0)) {
+                for (entry = hint->callbacks; entry;) {
+                    /* Save the next entry in case this one is deleted */
+                    SDL_HintWatch *next = entry->next;
+                    entry->callback(entry->userdata, name, hint->value, env);
+                    entry = next;
+                }
+            }
+            SDL_free(hint->value);
+            hint->value = NULL;
+            hint->priority = SDL_HINT_DEFAULT;
+            return SDL_TRUE;
+        }
+    }
+    return SDL_FALSE;
+}
+
+void SDL_ResetHints(void)
+{
+    const char *env;
+    SDL_Hint *hint;
+    SDL_HintWatch *entry;
+
+    for (hint = SDL_hints; hint; hint = hint->next) {
+        env = SDL_getenv(hint->name);
+        if ((env == NULL && hint->value != NULL) ||
+            (env != NULL && hint->value == NULL) ||
+            (env != NULL && SDL_strcmp(env, hint->value) != 0)) {
+            for (entry = hint->callbacks; entry;) {
+                /* Save the next entry in case this one is deleted */
+                SDL_HintWatch *next = entry->next;
+                entry->callback(entry->userdata, hint->name, hint->value, env);
+                entry = next;
+            }
+        }
+        SDL_free(hint->value);
+        hint->value = NULL;
+        hint->priority = SDL_HINT_DEFAULT;
+    }
+}
+
+SDL_bool SDL_SetHint(const char *name, const char *value)
+{
+    return SDL_SetHintWithPriority(name, value, SDL_HINT_NORMAL);
+}
+
+const char *SDL_GetHint(const char *name)
+{
+    const char *env;
+    SDL_Hint *hint;
+
+    env = SDL_getenv(name);
+    for (hint = SDL_hints; hint; hint = hint->next) {
+        if (SDL_strcmp(name, hint->name) == 0) {
+            if (env == NULL || hint->priority == SDL_HINT_OVERRIDE) {
+                return hint->value;
+            }
+            break;
+        }
+    }
+    return env;
+}
+
+SDL_bool SDL_GetStringBoolean(const char *value, SDL_bool default_value)
+{
+    if (value == NULL || !*value) {
+        return default_value;
+    }
+    if (*value == '0' || SDL_strcasecmp(value, "false") == 0) {
+        return SDL_FALSE;
+    }
+    return SDL_TRUE;
+}
+
+SDL_bool SDL_GetHintBoolean(const char *name, SDL_bool default_value)
+{
+    const char *hint = SDL_GetHint(name);
+    return SDL_GetStringBoolean(hint, default_value);
+}
+
+void SDL_AddHintCallback(const char *name, SDL_HintCallback callback, void *userdata)
+{
+    SDL_Hint *hint;
+    SDL_HintWatch *entry;
+    const char *value;
+
+    if (name == NULL || !*name) {
+        SDL_InvalidParamError("name");
+        return;
+    }
+    if (!callback) {
+        SDL_InvalidParamError("callback");
+        return;
+    }
+
+    SDL_DelHintCallback(name, callback, userdata);
+
+    entry = (SDL_HintWatch *)SDL_malloc(sizeof(*entry));
+    if (entry == NULL) {
+        SDL_OutOfMemory();
+        return;
+    }
+    entry->callback = callback;
+    entry->userdata = userdata;
+
+    for (hint = SDL_hints; hint; hint = hint->next) {
+        if (SDL_strcmp(name, hint->name) == 0) {
+            break;
+        }
+    }
+    if (hint == NULL) {
+        /* Need to add a hint entry for this watcher */
+        hint = (SDL_Hint *)SDL_malloc(sizeof(*hint));
+        if (hint == NULL) {
+            SDL_OutOfMemory();
+            SDL_free(entry);
+            return;
+        }
+        hint->name = SDL_strdup(name);
+        if (!hint->name) {
+            SDL_free(entry);
+            SDL_free(hint);
+            SDL_OutOfMemory();
+            return;
+        }
+        hint->value = NULL;
+        hint->priority = SDL_HINT_DEFAULT;
+        hint->callbacks = NULL;
+        hint->next = SDL_hints;
+        SDL_hints = hint;
+    }
+
+    /* Add it to the callbacks for this hint */
+    entry->next = hint->callbacks;
+    hint->callbacks = entry;
+
+    /* Now call it with the current value */
+    value = SDL_GetHint(name);
+    callback(userdata, name, value, value);
+}
+
+void SDL_DelHintCallback(const char *name, SDL_HintCallback callback, void *userdata)
+{
+    SDL_Hint *hint;
+    SDL_HintWatch *entry, *prev;
+
+    for (hint = SDL_hints; hint; hint = hint->next) {
+        if (SDL_strcmp(name, hint->name) == 0) {
+            prev = NULL;
+            for (entry = hint->callbacks; entry; entry = entry->next) {
+                if (callback == entry->callback && userdata == entry->userdata) {
+                    if (prev) {
+                        prev->next = entry->next;
+                    } else {
+                        hint->callbacks = entry->next;
+                    }
+                    SDL_free(entry);
+                    break;
+                }
+                prev = entry;
+            }
+            return;
+        }
+    }
+}
+
+void SDL_ClearHints(void)
+{
+    SDL_Hint *hint;
+    SDL_HintWatch *entry;
+
+    while (SDL_hints) {
+        hint = SDL_hints;
+        SDL_hints = hint->next;
+
+        SDL_free(hint->name);
+        SDL_free(hint->value);
+        for (entry = hint->callbacks; entry;) {
+            SDL_HintWatch *freeable = entry;
+            entry = entry->next;
+            SDL_free(freeable);
+        }
+        SDL_free(hint);
+    }
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_hints_c.h

@@ -0,0 +1,32 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+/* This file defines useful function for working with SDL hints */
+
+#ifndef SDL_hints_c_h_
+#define SDL_hints_c_h_
+
+extern SDL_bool SDL_GetStringBoolean(const char *value, SDL_bool default_value);
+
+#endif /* SDL_hints_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_internal.h

@@ -0,0 +1,133 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#ifndef SDL_internal_h_
+#define SDL_internal_h_
+
+/* Many of SDL's features require _GNU_SOURCE on various platforms */
+#ifndef _GNU_SOURCE
+#define _GNU_SOURCE
+#endif
+
+/* This is for a variable-length array at the end of a struct:
+    struct x { int y; char z[SDL_VARIABLE_LENGTH_ARRAY]; };
+   Use this because GCC 2 needs different magic than other compilers. */
+#if (defined(__GNUC__) && (__GNUC__ <= 2)) || defined(__CC_ARM) || defined(__cplusplus)
+#define SDL_VARIABLE_LENGTH_ARRAY 1
+#else
+#define SDL_VARIABLE_LENGTH_ARRAY
+#endif
+
+#define SDL_MAX_SMALL_ALLOC_STACKSIZE          128
+#define SDL_small_alloc(type, count, pisstack) ((*(pisstack) = ((sizeof(type) * (count)) < SDL_MAX_SMALL_ALLOC_STACKSIZE)), (*(pisstack) ? SDL_stack_alloc(type, count) : (type *)SDL_malloc(sizeof(type) * (count))))
+#define SDL_small_free(ptr, isstack) \
+    if ((isstack)) {                 \
+        SDL_stack_free(ptr);         \
+    } else {                         \
+        SDL_free(ptr);               \
+    }
+
+#include "dynapi/SDL_dynapi.h"
+
+#if SDL_DYNAMIC_API
+#include "dynapi/SDL_dynapi_overrides.h"
+/* force DECLSPEC off...it's all internal symbols now.
+   These will have actual #defines during SDL_dynapi.c only */
+#define DECLSPEC
+#endif
+
+#include "SDL_config.h"
+
+/* If you run into a warning that O_CLOEXEC is redefined, update the SDL configuration header for your platform to add HAVE_O_CLOEXEC */
+#ifndef HAVE_O_CLOEXEC
+#define O_CLOEXEC 0
+#endif
+
+/* A few #defines to reduce SDL2 footprint.
+   Only effective when library is statically linked.
+   You have to manually edit this file. */
+#ifndef SDL_LEAN_AND_MEAN
+#define SDL_LEAN_AND_MEAN 0
+#endif
+
+/* Optimized functions from 'SDL_blit_0.c'
+   - blit with source BitsPerPixel < 8, palette */
+#ifndef SDL_HAVE_BLIT_0
+#define SDL_HAVE_BLIT_0 !SDL_LEAN_AND_MEAN
+#endif
+
+/* Optimized functions from 'SDL_blit_1.c'
+   - blit with source BytesPerPixel == 1, palette */
+#ifndef SDL_HAVE_BLIT_1
+#define SDL_HAVE_BLIT_1 !SDL_LEAN_AND_MEAN
+#endif
+
+/* Optimized functions from 'SDL_blit_A.c'
+   - blit with 'SDL_BLENDMODE_BLEND' blending mode */
+#ifndef SDL_HAVE_BLIT_A
+#define SDL_HAVE_BLIT_A !SDL_LEAN_AND_MEAN
+#endif
+
+/* Optimized functions from 'SDL_blit_N.c'
+   - blit with COLORKEY mode, or nothing */
+#ifndef SDL_HAVE_BLIT_N
+#define SDL_HAVE_BLIT_N !SDL_LEAN_AND_MEAN
+#endif
+
+/* Optimized functions from 'SDL_blit_N.c'
+   - RGB565 conversion with Lookup tables */
+#ifndef SDL_HAVE_BLIT_N_RGB565
+#define SDL_HAVE_BLIT_N_RGB565 !SDL_LEAN_AND_MEAN
+#endif
+
+/* Optimized functions from 'SDL_blit_AUTO.c'
+   - blit with modulate color, modulate alpha, any blending mode
+   - scaling or not */
+#ifndef SDL_HAVE_BLIT_AUTO
+#define SDL_HAVE_BLIT_AUTO !SDL_LEAN_AND_MEAN
+#endif
+
+/* Run-Length-Encoding
+   - SDL_SetColorKey() called with SDL_RLEACCEL flag */
+#ifndef SDL_HAVE_RLE
+#define SDL_HAVE_RLE !SDL_LEAN_AND_MEAN
+#endif
+
+/* Software SDL_Renderer
+   - creation of software renderer
+   - *not* general blitting functions
+   - {blend,draw}{fillrect,line,point} internal functions */
+#ifndef SDL_VIDEO_RENDER_SW
+#define SDL_VIDEO_RENDER_SW !SDL_LEAN_AND_MEAN
+#endif
+
+/* YUV formats
+   - handling of YUV surfaces
+   - blitting and conversion functions */
+#ifndef SDL_HAVE_YUV
+#define SDL_HAVE_YUV !SDL_LEAN_AND_MEAN
+#endif
+
+#include "SDL_assert.h"
+#include "SDL_log.h"
+
+#endif /* SDL_internal_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_list.c

@@ -0,0 +1,89 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+#include "SDL.h"
+#include "./SDL_list.h"
+
+/* Push */
+int SDL_ListAdd(SDL_ListNode **head, void *ent)
+{
+    SDL_ListNode *node = SDL_malloc(sizeof(*node));
+
+    if (node == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    node->entry = ent;
+    node->next = *head;
+    *head = node;
+    return 0;
+}
+
+/* Pop from end as a FIFO (if add with SDL_ListAdd) */
+void SDL_ListPop(SDL_ListNode **head, void **ent)
+{
+    SDL_ListNode **ptr = head;
+
+    /* Invalid or empty */
+    if (head == NULL || *head == NULL) {
+        return;
+    }
+
+    while ((*ptr)->next) {
+        ptr = &(*ptr)->next;
+    }
+
+    if (ent) {
+        *ent = (*ptr)->entry;
+    }
+
+    SDL_free(*ptr);
+    *ptr = NULL;
+}
+
+void SDL_ListRemove(SDL_ListNode **head, void *ent)
+{
+    SDL_ListNode **ptr = head;
+
+    while (*ptr) {
+        if ((*ptr)->entry == ent) {
+            SDL_ListNode *tmp = *ptr;
+            *ptr = (*ptr)->next;
+            SDL_free(tmp);
+            return;
+        }
+        ptr = &(*ptr)->next;
+    }
+}
+
+void SDL_ListClear(SDL_ListNode **head)
+{
+    SDL_ListNode *l = *head;
+    *head = NULL;
+    while (l) {
+        SDL_ListNode *tmp = l;
+        l = l->next;
+        SDL_free(tmp);
+    }
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_list.h

@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_list_h_
+#define SDL_list_h_
+
+typedef struct SDL_ListNode
+{
+    void *entry;
+    struct SDL_ListNode *next;
+} SDL_ListNode;
+
+int SDL_ListAdd(SDL_ListNode **head, void *ent);
+void SDL_ListPop(SDL_ListNode **head, void **ent);
+void SDL_ListRemove(SDL_ListNode **head, void *ent);
+void SDL_ListClear(SDL_ListNode **head);
+
+#endif /* SDL_list_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_log.c

@@ -0,0 +1,505 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+#if defined(__WIN32__) || defined(__WINRT__) || defined(__GDK__)
+#include "core/windows/SDL_windows.h"
+#endif
+
+/* Simple log messages in SDL */
+
+#include "SDL_error.h"
+#include "SDL_log.h"
+#include "SDL_mutex.h"
+#include "SDL_log_c.h"
+
+#if HAVE_STDIO_H
+#include <stdio.h>
+#endif
+
+#if defined(__ANDROID__)
+#include <android/log.h>
+#endif
+
+#include "stdlib/SDL_vacopy.h"
+
+/* The size of the stack buffer to use for rendering log messages. */
+#define SDL_MAX_LOG_MESSAGE_STACK 256
+
+#define DEFAULT_PRIORITY             SDL_LOG_PRIORITY_CRITICAL
+#define DEFAULT_ASSERT_PRIORITY      SDL_LOG_PRIORITY_WARN
+#define DEFAULT_APPLICATION_PRIORITY SDL_LOG_PRIORITY_INFO
+#define DEFAULT_TEST_PRIORITY        SDL_LOG_PRIORITY_VERBOSE
+
+typedef struct SDL_LogLevel
+{
+    int category;
+    SDL_LogPriority priority;
+    struct SDL_LogLevel *next;
+} SDL_LogLevel;
+
+/* The default log output function */
+static void SDLCALL SDL_LogOutput(void *userdata, int category, SDL_LogPriority priority, const char *message);
+
+static SDL_LogLevel *SDL_loglevels;
+static SDL_LogPriority SDL_default_priority = DEFAULT_PRIORITY;
+static SDL_LogPriority SDL_assert_priority = DEFAULT_ASSERT_PRIORITY;
+static SDL_LogPriority SDL_application_priority = DEFAULT_APPLICATION_PRIORITY;
+static SDL_LogPriority SDL_test_priority = DEFAULT_TEST_PRIORITY;
+static SDL_LogOutputFunction SDL_log_function = SDL_LogOutput;
+static void *SDL_log_userdata = NULL;
+static SDL_mutex *log_function_mutex = NULL;
+
+static const char *SDL_priority_prefixes[SDL_NUM_LOG_PRIORITIES] = {
+    NULL,
+    "VERBOSE",
+    "DEBUG",
+    "INFO",
+    "WARN",
+    "ERROR",
+    "CRITICAL"
+};
+
+#ifdef __ANDROID__
+static const char *SDL_category_prefixes[] = {
+    "APP",
+    "ERROR",
+    "ASSERT",
+    "SYSTEM",
+    "AUDIO",
+    "VIDEO",
+    "RENDER",
+    "INPUT",
+    "TEST"
+};
+
+SDL_COMPILE_TIME_ASSERT(category_prefixes_enum, SDL_TABLESIZE(SDL_category_prefixes) == SDL_LOG_CATEGORY_RESERVED1);
+
+static int SDL_android_priority[SDL_NUM_LOG_PRIORITIES] = {
+    ANDROID_LOG_UNKNOWN,
+    ANDROID_LOG_VERBOSE,
+    ANDROID_LOG_DEBUG,
+    ANDROID_LOG_INFO,
+    ANDROID_LOG_WARN,
+    ANDROID_LOG_ERROR,
+    ANDROID_LOG_FATAL
+};
+#endif /* __ANDROID__ */
+
+void SDL_LogInit(void)
+{
+    if (log_function_mutex == NULL) {
+        /* if this fails we'll try to continue without it. */
+        log_function_mutex = SDL_CreateMutex();
+    }
+}
+
+void SDL_LogQuit(void)
+{
+    SDL_LogResetPriorities();
+    if (log_function_mutex) {
+        SDL_DestroyMutex(log_function_mutex);
+        log_function_mutex = NULL;
+    }
+}
+
+void SDL_LogSetAllPriority(SDL_LogPriority priority)
+{
+    SDL_LogLevel *entry;
+
+    for (entry = SDL_loglevels; entry; entry = entry->next) {
+        entry->priority = priority;
+    }
+    SDL_default_priority = priority;
+    SDL_assert_priority = priority;
+    SDL_application_priority = priority;
+}
+
+void SDL_LogSetPriority(int category, SDL_LogPriority priority)
+{
+    SDL_LogLevel *entry;
+
+    for (entry = SDL_loglevels; entry; entry = entry->next) {
+        if (entry->category == category) {
+            entry->priority = priority;
+            return;
+        }
+    }
+
+    /* Create a new entry */
+    entry = (SDL_LogLevel *)SDL_malloc(sizeof(*entry));
+    if (entry) {
+        entry->category = category;
+        entry->priority = priority;
+        entry->next = SDL_loglevels;
+        SDL_loglevels = entry;
+    }
+}
+
+SDL_LogPriority SDL_LogGetPriority(int category)
+{
+    SDL_LogLevel *entry;
+
+    for (entry = SDL_loglevels; entry; entry = entry->next) {
+        if (entry->category == category) {
+            return entry->priority;
+        }
+    }
+
+    if (category == SDL_LOG_CATEGORY_TEST) {
+        return SDL_test_priority;
+    } else if (category == SDL_LOG_CATEGORY_APPLICATION) {
+        return SDL_application_priority;
+    } else if (category == SDL_LOG_CATEGORY_ASSERT) {
+        return SDL_assert_priority;
+    } else {
+        return SDL_default_priority;
+    }
+}
+
+void SDL_LogResetPriorities(void)
+{
+    SDL_LogLevel *entry;
+
+    while (SDL_loglevels) {
+        entry = SDL_loglevels;
+        SDL_loglevels = entry->next;
+        SDL_free(entry);
+    }
+
+    SDL_default_priority = DEFAULT_PRIORITY;
+    SDL_assert_priority = DEFAULT_ASSERT_PRIORITY;
+    SDL_application_priority = DEFAULT_APPLICATION_PRIORITY;
+    SDL_test_priority = DEFAULT_TEST_PRIORITY;
+}
+
+void SDL_Log(SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(SDL_LOG_CATEGORY_APPLICATION, SDL_LOG_PRIORITY_INFO, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogVerbose(int category, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, SDL_LOG_PRIORITY_VERBOSE, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogDebug(int category, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, SDL_LOG_PRIORITY_DEBUG, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogInfo(int category, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, SDL_LOG_PRIORITY_INFO, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogWarn(int category, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, SDL_LOG_PRIORITY_WARN, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogError(int category, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, SDL_LOG_PRIORITY_ERROR, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogCritical(int category, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, SDL_LOG_PRIORITY_CRITICAL, fmt, ap);
+    va_end(ap);
+}
+
+void SDL_LogMessage(int category, SDL_LogPriority priority, SDL_PRINTF_FORMAT_STRING const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start(ap, fmt);
+    SDL_LogMessageV(category, priority, fmt, ap);
+    va_end(ap);
+}
+
+#ifdef __ANDROID__
+static const char *GetCategoryPrefix(int category)
+{
+    if (category < SDL_LOG_CATEGORY_RESERVED1) {
+        return SDL_category_prefixes[category];
+    }
+    if (category < SDL_LOG_CATEGORY_CUSTOM) {
+        return "RESERVED";
+    }
+    return "CUSTOM";
+}
+#endif /* __ANDROID__ */
+
+void SDL_LogMessageV(int category, SDL_LogPriority priority, const char *fmt, va_list ap)
+{
+    char *message = NULL;
+    char stack_buf[SDL_MAX_LOG_MESSAGE_STACK];
+    size_t len_plus_term;
+    int len;
+    va_list aq;
+
+    /* Nothing to do if we don't have an output function */
+    if (!SDL_log_function) {
+        return;
+    }
+
+    /* Make sure we don't exceed array bounds */
+    if ((int)priority < 0 || priority >= SDL_NUM_LOG_PRIORITIES) {
+        return;
+    }
+
+    /* See if we want to do anything with this message */
+    if (priority < SDL_LogGetPriority(category)) {
+        return;
+    }
+
+    if (log_function_mutex == NULL) {
+        /* this mutex creation can race if you log from two threads at startup. You should have called SDL_Init first! */
+        log_function_mutex = SDL_CreateMutex();
+    }
+
+    /* Render into stack buffer */
+    va_copy(aq, ap);
+    len = SDL_vsnprintf(stack_buf, sizeof(stack_buf), fmt, aq);
+    va_end(aq);
+
+    if (len < 0) {
+        return;
+    }
+
+    /* If message truncated, allocate and re-render */
+    if (len >= sizeof(stack_buf) && SDL_size_add_overflow(len, 1, &len_plus_term) == 0) {
+        /* Allocate exactly what we need, including the zero-terminator */
+        message = (char *)SDL_malloc(len_plus_term);
+        if (message == NULL) {
+            return;
+        }
+        va_copy(aq, ap);
+        len = SDL_vsnprintf(message, len_plus_term, fmt, aq);
+        va_end(aq);
+    } else {
+        message = stack_buf;
+    }
+
+    /* Chop off final endline. */
+    if ((len > 0) && (message[len - 1] == '\n')) {
+        message[--len] = '\0';
+        if ((len > 0) && (message[len - 1] == '\r')) { /* catch "\r\n", too. */
+            message[--len] = '\0';
+        }
+    }
+
+    SDL_LockMutex(log_function_mutex);
+    SDL_log_function(SDL_log_userdata, category, priority, message);
+    SDL_UnlockMutex(log_function_mutex);
+
+    /* Free only if dynamically allocated */
+    if (message != stack_buf) {
+        SDL_free(message);
+    }
+}
+
+#if defined(__WIN32__) && !defined(HAVE_STDIO_H) && !defined(__WINRT__) && !defined(__GDK__)
+/* Flag tracking the attachment of the console: 0=unattached, 1=attached to a console, 2=attached to a file, -1=error */
+static int consoleAttached = 0;
+
+/* Handle to stderr output of console. */
+static HANDLE stderrHandle = NULL;
+#endif
+
+static void SDLCALL SDL_LogOutput(void *userdata, int category, SDL_LogPriority priority,
+                                  const char *message)
+{
+#if defined(__WIN32__) || defined(__WINRT__) || defined(__GDK__)
+    /* Way too many allocations here, urgh */
+    /* Note: One can't call SDL_SetError here, since that function itself logs. */
+    {
+        char *output;
+        size_t length;
+        LPTSTR tstr;
+        SDL_bool isstack;
+
+#if !defined(HAVE_STDIO_H) && !defined(__WINRT__) && !defined(__GDK__)
+        BOOL attachResult;
+        DWORD attachError;
+        DWORD charsWritten;
+        DWORD consoleMode;
+
+        /* Maybe attach console and get stderr handle */
+        if (consoleAttached == 0) {
+            attachResult = AttachConsole(ATTACH_PARENT_PROCESS);
+            if (!attachResult) {
+                attachError = GetLastError();
+                if (attachError == ERROR_INVALID_HANDLE) {
+                    /* This is expected when running from Visual Studio */
+                    /*OutputDebugString(TEXT("Parent process has no console\r\n"));*/
+                    consoleAttached = -1;
+                } else if (attachError == ERROR_GEN_FAILURE) {
+                    OutputDebugString(TEXT("Could not attach to console of parent process\r\n"));
+                    consoleAttached = -1;
+                } else if (attachError == ERROR_ACCESS_DENIED) {
+                    /* Already attached */
+                    consoleAttached = 1;
+                } else {
+                    OutputDebugString(TEXT("Error attaching console\r\n"));
+                    consoleAttached = -1;
+                }
+            } else {
+                /* Newly attached */
+                consoleAttached = 1;
+            }
+
+            if (consoleAttached == 1) {
+                stderrHandle = GetStdHandle(STD_ERROR_HANDLE);
+
+                if (GetConsoleMode(stderrHandle, &consoleMode) == 0) {
+                    /* WriteConsole fails if the output is redirected to a file. Must use WriteFile instead. */
+                    consoleAttached = 2;
+                }
+            }
+        }
+#endif /* !defined(HAVE_STDIO_H) && !defined(__WINRT__) && !defined(__GDK__) */
+
+        length = SDL_strlen(SDL_priority_prefixes[priority]) + 2 + SDL_strlen(message) + 1 + 1 + 1;
+        output = SDL_small_alloc(char, length, &isstack);
+        (void)SDL_snprintf(output, length, "%s: %s\r\n", SDL_priority_prefixes[priority], message);
+        tstr = WIN_UTF8ToString(output);
+
+        /* Output to debugger */
+        OutputDebugString(tstr);
+
+#if !defined(HAVE_STDIO_H) && !defined(__WINRT__) && !defined(__GDK__)
+        /* Screen output to stderr, if console was attached. */
+        if (consoleAttached == 1) {
+            if (!WriteConsole(stderrHandle, tstr, (DWORD)SDL_tcslen(tstr), &charsWritten, NULL)) {
+                OutputDebugString(TEXT("Error calling WriteConsole\r\n"));
+                if (GetLastError() == ERROR_NOT_ENOUGH_MEMORY) {
+                    OutputDebugString(TEXT("Insufficient heap memory to write message\r\n"));
+                }
+            }
+
+        } else if (consoleAttached == 2) {
+            if (!WriteFile(stderrHandle, output, (DWORD)SDL_strlen(output), &charsWritten, NULL)) {
+                OutputDebugString(TEXT("Error calling WriteFile\r\n"));
+            }
+        }
+#endif /* !defined(HAVE_STDIO_H) && !defined(__WINRT__) && !defined(__GDK__) */
+
+        SDL_free(tstr);
+        SDL_small_free(output, isstack);
+    }
+#elif defined(__ANDROID__)
+    {
+        char tag[32];
+
+        SDL_snprintf(tag, SDL_arraysize(tag), "SDL/%s", GetCategoryPrefix(category));
+        __android_log_write(SDL_android_priority[priority], tag, message);
+    }
+#elif defined(__APPLE__) && (defined(SDL_VIDEO_DRIVER_COCOA) || defined(SDL_VIDEO_DRIVER_UIKIT))
+    /* Technically we don't need Cocoa/UIKit, but that's where this function is defined for now.
+     */
+    extern void SDL_NSLog(const char *prefix, const char *text);
+    {
+        SDL_NSLog(SDL_priority_prefixes[priority], message);
+        return;
+    }
+#elif defined(__PSP__) || defined(__PS2__)
+    {
+        FILE *pFile;
+        pFile = fopen("SDL_Log.txt", "a");
+        if (pFile != NULL) {
+            (void)fprintf(pFile, "%s: %s\n", SDL_priority_prefixes[priority], message);
+            (void)fclose(pFile);
+        }
+    }
+#elif defined(__VITA__)
+    {
+        FILE *pFile;
+        pFile = fopen("ux0:/data/SDL_Log.txt", "a");
+        if (pFile != NULL) {
+            (void)fprintf(pFile, "%s: %s\n", SDL_priority_prefixes[priority], message);
+            (void)fclose(pFile);
+        }
+    }
+#elif defined(__3DS__)
+    {
+        FILE *pFile;
+        pFile = fopen("sdmc:/3ds/SDL_Log.txt", "a");
+        if (pFile != NULL) {
+            (void)fprintf(pFile, "%s: %s\n", SDL_priority_prefixes[priority], message);
+            (void)fclose(pFile);
+        }
+    }
+#endif
+#if HAVE_STDIO_H && \
+    !(defined(__APPLE__) && (defined(SDL_VIDEO_DRIVER_COCOA) || defined(SDL_VIDEO_DRIVER_UIKIT)))
+    fprintf(stderr, "%s: %s\n", SDL_priority_prefixes[priority], message);
+#if __NACL__
+    fflush(stderr);
+#endif
+#endif
+}
+
+void SDL_LogGetOutputFunction(SDL_LogOutputFunction *callback, void **userdata)
+{
+    if (callback) {
+        *callback = SDL_log_function;
+    }
+    if (userdata) {
+        *userdata = SDL_log_userdata;
+    }
+}
+
+void SDL_LogSetOutputFunction(SDL_LogOutputFunction callback, void *userdata)
+{
+    SDL_log_function = callback;
+    SDL_log_userdata = userdata;
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_log_c.h

@@ -0,0 +1,33 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "./SDL_internal.h"
+
+/* This file defines useful function for working with SDL logging */
+
+#ifndef SDL_log_c_h_
+#define SDL_log_c_h_
+
+extern void SDL_LogInit(void);
+extern void SDL_LogQuit(void);
+
+#endif /* SDL_log_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_utils.c

@@ -0,0 +1,52 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#include "SDL_utils_c.h"
+
+/* Common utility functions that aren't in the public API */
+
+int SDL_powerof2(int x)
+{
+    int value;
+
+    if (x <= 0) {
+        /* Return some sane value - we shouldn't hit this in our use cases */
+        return 1;
+    }
+
+    /* This trick works for 32-bit values */
+    {
+        SDL_COMPILE_TIME_ASSERT(SDL_powerof2, sizeof(x) == sizeof(Uint32));
+    }
+    value = x;
+    value -= 1;
+    value |= value >> 1;
+    value |= value >> 2;
+    value |= value >> 4;
+    value |= value >> 8;
+    value |= value >> 16;
+    value += 1;
+
+    return value;
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/SDL_utils_c.h

@@ -0,0 +1,32 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_utils_h_
+#define SDL_utils_h_
+
+/* Common utility functions that aren't in the public API */
+
+/* Return the smallest power of 2 greater than or equal to 'x' */
+extern int SDL_powerof2(int x);
+
+#endif /* SDL_utils_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/atomic/SDL_atomic.c

@@ -0,0 +1,305 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+#include "SDL_atomic.h"
+
+#if defined(_MSC_VER) && (_MSC_VER >= 1500)
+#include <intrin.h>
+#define HAVE_MSC_ATOMICS 1
+#endif
+
+#if defined(__MACOSX__)  /* !!! FIXME: should we favor gcc atomics? */
+#include <libkern/OSAtomic.h>
+#endif
+
+#if !defined(HAVE_GCC_ATOMICS) && defined(__SOLARIS__)
+#include <atomic.h>
+#endif
+
+/* The __atomic_load_n() intrinsic showed up in different times for different compilers. */
+#if defined(__clang__)
+#if __has_builtin(__atomic_load_n) || defined(HAVE_GCC_ATOMICS)
+/* !!! FIXME: this advertises as available in the NDK but uses an external symbol we don't have.
+   It might be in a later NDK or we might need an extra library? --ryan. */
+#if !defined(__ANDROID__)
+#define HAVE_ATOMIC_LOAD_N 1
+#endif
+#endif
+#elif defined(__GNUC__)
+#if (__GNUC__ >= 5)
+#define HAVE_ATOMIC_LOAD_N 1
+#endif
+#endif
+
+/* *INDENT-OFF* */ /* clang-format off */
+#if defined(__WATCOMC__) && defined(__386__)
+SDL_COMPILE_TIME_ASSERT(intsize, 4==sizeof(int));
+#define HAVE_WATCOM_ATOMICS
+extern __inline int _SDL_xchg_watcom(volatile int *a, int v);
+#pragma aux _SDL_xchg_watcom = \
+  "lock xchg [ecx], eax" \
+  parm [ecx] [eax] \
+  value [eax] \
+  modify exact [eax];
+
+extern __inline unsigned char _SDL_cmpxchg_watcom(volatile int *a, int newval, int oldval);
+#pragma aux _SDL_cmpxchg_watcom = \
+  "lock cmpxchg [edx], ecx" \
+  "setz al" \
+  parm [edx] [ecx] [eax] \
+  value [al] \
+  modify exact [eax];
+
+extern __inline int _SDL_xadd_watcom(volatile int *a, int v);
+#pragma aux _SDL_xadd_watcom = \
+  "lock xadd [ecx], eax" \
+  parm [ecx] [eax] \
+  value [eax] \
+  modify exact [eax];
+
+#endif /* __WATCOMC__ && __386__ */
+/* *INDENT-ON* */ /* clang-format on */
+
+/*
+  If any of the operations are not provided then we must emulate some
+  of them. That means we need a nice implementation of spin locks
+  that avoids the "one big lock" problem. We use a vector of spin
+  locks and pick which one to use based on the address of the operand
+  of the function.
+
+  To generate the index of the lock we first shift by 3 bits to get
+  rid on the zero bits that result from 32 and 64 bit allignment of
+  data. We then mask off all but 5 bits and use those 5 bits as an
+  index into the table.
+
+  Picking the lock this way insures that accesses to the same data at
+  the same time will go to the same lock. OTOH, accesses to different
+  data have only a 1/32 chance of hitting the same lock. That should
+  pretty much eliminate the chances of several atomic operations on
+  different data from waiting on the same "big lock". If it isn't
+  then the table of locks can be expanded to a new size so long as
+  the new size is a power of two.
+
+  Contributed by Bob Pendleton, bob@pendleton.com
+*/
+
+#if !defined(HAVE_MSC_ATOMICS) && !defined(HAVE_GCC_ATOMICS) && !defined(__MACOSX__) && !defined(__SOLARIS__) && !defined(HAVE_WATCOM_ATOMICS)
+#define EMULATE_CAS 1
+#endif
+
+#if EMULATE_CAS
+static SDL_SpinLock locks[32];
+
+static SDL_INLINE void enterLock(void *a)
+{
+    uintptr_t index = ((((uintptr_t)a) >> 3) & 0x1f);
+
+    SDL_AtomicLock(&locks[index]);
+}
+
+static SDL_INLINE void leaveLock(void *a)
+{
+    uintptr_t index = ((((uintptr_t)a) >> 3) & 0x1f);
+
+    SDL_AtomicUnlock(&locks[index]);
+}
+#endif
+
+SDL_bool SDL_AtomicCAS(SDL_atomic_t *a, int oldval, int newval)
+{
+#ifdef HAVE_MSC_ATOMICS
+    SDL_COMPILE_TIME_ASSERT(atomic_cas, sizeof(long) == sizeof(a->value));
+    return _InterlockedCompareExchange((long *)&a->value, (long)newval, (long)oldval) == (long)oldval;
+#elif defined(HAVE_WATCOM_ATOMICS)
+    return (SDL_bool)_SDL_cmpxchg_watcom(&a->value, newval, oldval);
+#elif defined(HAVE_GCC_ATOMICS)
+    return (SDL_bool) __sync_bool_compare_and_swap(&a->value, oldval, newval);
+#elif defined(__MACOSX__)  /* this is deprecated in 10.12 sdk; favor gcc atomics. */
+    return (SDL_bool) OSAtomicCompareAndSwap32Barrier(oldval, newval, &a->value);
+#elif defined(__SOLARIS__)
+    return (SDL_bool)((int)atomic_cas_uint((volatile uint_t *)&a->value, (uint_t)oldval, (uint_t)newval) == oldval);
+#elif EMULATE_CAS
+    SDL_bool retval = SDL_FALSE;
+
+    enterLock(a);
+    if (a->value == oldval) {
+        a->value = newval;
+        retval = SDL_TRUE;
+    }
+    leaveLock(a);
+
+    return retval;
+#else
+#error Please define your platform.
+#endif
+}
+
+SDL_bool SDL_AtomicCASPtr(void **a, void *oldval, void *newval)
+{
+#if defined(HAVE_MSC_ATOMICS)
+    return _InterlockedCompareExchangePointer(a, newval, oldval) == oldval;
+#elif defined(HAVE_WATCOM_ATOMICS)
+    return (SDL_bool)_SDL_cmpxchg_watcom((int *)a, (long)newval, (long)oldval);
+#elif defined(HAVE_GCC_ATOMICS)
+    return __sync_bool_compare_and_swap(a, oldval, newval);
+#elif defined(__MACOSX__) && defined(__LP64__)  /* this is deprecated in 10.12 sdk; favor gcc atomics. */
+    return (SDL_bool) OSAtomicCompareAndSwap64Barrier((int64_t)oldval, (int64_t)newval, (int64_t*) a);
+#elif defined(__MACOSX__) && !defined(__LP64__)  /* this is deprecated in 10.12 sdk; favor gcc atomics. */
+    return (SDL_bool) OSAtomicCompareAndSwap32Barrier((int32_t)oldval, (int32_t)newval, (int32_t*) a);
+#elif defined(__SOLARIS__)
+    return (SDL_bool)(atomic_cas_ptr(a, oldval, newval) == oldval);
+#elif EMULATE_CAS
+    SDL_bool retval = SDL_FALSE;
+
+    enterLock(a);
+    if (*a == oldval) {
+        *a = newval;
+        retval = SDL_TRUE;
+    }
+    leaveLock(a);
+
+    return retval;
+#else
+#error Please define your platform.
+#endif
+}
+
+int SDL_AtomicSet(SDL_atomic_t *a, int v)
+{
+#ifdef HAVE_MSC_ATOMICS
+    SDL_COMPILE_TIME_ASSERT(atomic_set, sizeof(long) == sizeof(a->value));
+    return _InterlockedExchange((long *)&a->value, v);
+#elif defined(HAVE_WATCOM_ATOMICS)
+    return _SDL_xchg_watcom(&a->value, v);
+#elif defined(HAVE_GCC_ATOMICS)
+    return __sync_lock_test_and_set(&a->value, v);
+#elif defined(__SOLARIS__)
+    return (int)atomic_swap_uint((volatile uint_t *)&a->value, v);
+#else
+    int value;
+    do {
+        value = a->value;
+    } while (!SDL_AtomicCAS(a, value, v));
+    return value;
+#endif
+}
+
+void *SDL_AtomicSetPtr(void **a, void *v)
+{
+#if defined(HAVE_MSC_ATOMICS)
+    return _InterlockedExchangePointer(a, v);
+#elif defined(HAVE_WATCOM_ATOMICS)
+    return (void *)_SDL_xchg_watcom((int *)a, (long)v);
+#elif defined(HAVE_GCC_ATOMICS)
+    return __sync_lock_test_and_set(a, v);
+#elif defined(__SOLARIS__)
+    return atomic_swap_ptr(a, v);
+#else
+    void *value;
+    do {
+        value = *a;
+    } while (!SDL_AtomicCASPtr(a, value, v));
+    return value;
+#endif
+}
+
+int SDL_AtomicAdd(SDL_atomic_t *a, int v)
+{
+#ifdef HAVE_MSC_ATOMICS
+    SDL_COMPILE_TIME_ASSERT(atomic_add, sizeof(long) == sizeof(a->value));
+    return _InterlockedExchangeAdd((long *)&a->value, v);
+#elif defined(HAVE_WATCOM_ATOMICS)
+    return _SDL_xadd_watcom(&a->value, v);
+#elif defined(HAVE_GCC_ATOMICS)
+    return __sync_fetch_and_add(&a->value, v);
+#elif defined(__SOLARIS__)
+    int pv = a->value;
+    membar_consumer();
+    atomic_add_int((volatile uint_t *)&a->value, v);
+    return pv;
+#else
+    int value;
+    do {
+        value = a->value;
+    } while (!SDL_AtomicCAS(a, value, (value + v)));
+    return value;
+#endif
+}
+
+int SDL_AtomicGet(SDL_atomic_t *a)
+{
+#ifdef HAVE_ATOMIC_LOAD_N
+    return __atomic_load_n(&a->value, __ATOMIC_SEQ_CST);
+#elif defined(HAVE_MSC_ATOMICS)
+    SDL_COMPILE_TIME_ASSERT(atomic_get, sizeof(long) == sizeof(a->value));
+    return _InterlockedOr((long *)&a->value, 0);
+#elif defined(HAVE_WATCOM_ATOMICS)
+    return _SDL_xadd_watcom(&a->value, 0);
+#elif defined(HAVE_GCC_ATOMICS)
+    return __sync_or_and_fetch(&a->value, 0);
+#elif defined(__MACOSX__)  /* this is deprecated in 10.12 sdk; favor gcc atomics. */
+    return sizeof(a->value) == sizeof(uint32_t) ? OSAtomicOr32Barrier(0, (volatile uint32_t *)&a->value) : OSAtomicAdd64Barrier(0, (volatile int64_t *)&a->value);
+#elif defined(__SOLARIS__)
+    return atomic_or_uint((volatile uint_t *)&a->value, 0);
+#else
+    int value;
+    do {
+        value = a->value;
+    } while (!SDL_AtomicCAS(a, value, value));
+    return value;
+#endif
+}
+
+void *SDL_AtomicGetPtr(void **a)
+{
+#ifdef HAVE_ATOMIC_LOAD_N
+    return __atomic_load_n(a, __ATOMIC_SEQ_CST);
+#elif defined(HAVE_MSC_ATOMICS)
+    return _InterlockedCompareExchangePointer(a, NULL, NULL);
+#elif defined(HAVE_GCC_ATOMICS)
+    return __sync_val_compare_and_swap(a, (void *)0, (void *)0);
+#elif defined(__SOLARIS__)
+    return atomic_cas_ptr(a, (void *)0, (void *)0);
+#else
+    void *value;
+    do {
+        value = *a;
+    } while (!SDL_AtomicCASPtr(a, value, value));
+    return value;
+#endif
+}
+
+#ifdef SDL_MEMORY_BARRIER_USES_FUNCTION
+#error This file should be built in arm mode so the mcr instruction is available for memory barriers
+#endif
+
+void SDL_MemoryBarrierReleaseFunction(void)
+{
+    SDL_MemoryBarrierRelease();
+}
+
+void SDL_MemoryBarrierAcquireFunction(void)
+{
+    SDL_MemoryBarrierAcquire();
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/atomic/SDL_spinlock.c

@@ -0,0 +1,213 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+#if defined(__WIN32__) || defined(__WINRT__) || defined(__GDK__)
+#include "../core/windows/SDL_windows.h"
+#endif
+
+#include "SDL_atomic.h"
+#include "SDL_mutex.h"
+#include "SDL_timer.h"
+
+#if !defined(HAVE_GCC_ATOMICS) && defined(__SOLARIS__)
+#include <atomic.h>
+#endif
+
+#if !defined(HAVE_GCC_ATOMICS) && defined(__RISCOS__)
+#include <unixlib/local.h>
+#endif
+
+#if defined(_MSC_VER) && (defined(_M_IX86) || defined(_M_X64))
+#include <xmmintrin.h>
+#endif
+
+#if defined(PS2)
+#include <kernel.h>
+#endif
+
+#if !defined(HAVE_GCC_ATOMICS) && defined(__MACOSX__)
+#include <libkern/OSAtomic.h>
+#endif
+
+/* *INDENT-OFF* */ /* clang-format off */
+#if defined(__WATCOMC__) && defined(__386__)
+SDL_COMPILE_TIME_ASSERT(locksize, 4==sizeof(SDL_SpinLock));
+extern __inline int _SDL_xchg_watcom(volatile int *a, int v);
+#pragma aux _SDL_xchg_watcom = \
+  "lock xchg [ecx], eax" \
+  parm [ecx] [eax] \
+  value [eax] \
+  modify exact [eax];
+#endif /* __WATCOMC__ && __386__ */
+/* *INDENT-ON* */ /* clang-format on */
+
+/* This function is where all the magic happens... */
+SDL_bool SDL_AtomicTryLock(SDL_SpinLock *lock)
+{
+#if SDL_ATOMIC_DISABLED
+    /* Terrible terrible damage */
+    static SDL_mutex *_spinlock_mutex;
+
+    if (_spinlock_mutex == NULL) {
+        /* Race condition on first lock... */
+        _spinlock_mutex = SDL_CreateMutex();
+    }
+    SDL_LockMutex(_spinlock_mutex);
+    if (*lock == 0) {
+        *lock = 1;
+        SDL_UnlockMutex(_spinlock_mutex);
+        return SDL_TRUE;
+    } else {
+        SDL_UnlockMutex(_spinlock_mutex);
+        return SDL_FALSE;
+    }
+
+#elif HAVE_GCC_ATOMICS || HAVE_GCC_SYNC_LOCK_TEST_AND_SET
+    return __sync_lock_test_and_set(lock, 1) == 0;
+
+#elif defined(_MSC_VER) && (defined(_M_ARM) || defined(_M_ARM64))
+    return _InterlockedExchange_acq(lock, 1) == 0;
+
+#elif defined(_MSC_VER)
+    SDL_COMPILE_TIME_ASSERT(locksize, sizeof(*lock) == sizeof(long));
+    return InterlockedExchange((long *)lock, 1) == 0;
+
+#elif defined(__WATCOMC__) && defined(__386__)
+    return _SDL_xchg_watcom(lock, 1) == 0;
+
+#elif defined(__GNUC__) && defined(__arm__) &&               \
+    (defined(__ARM_ARCH_3__) || defined(__ARM_ARCH_3M__) ||  \
+     defined(__ARM_ARCH_4__) || defined(__ARM_ARCH_4T__) ||  \
+     defined(__ARM_ARCH_5__) || defined(__ARM_ARCH_5TE__) || \
+     defined(__ARM_ARCH_5TEJ__))
+    int result;
+
+#if defined(__RISCOS__)
+    if (__cpucap_have_rex()) {
+        __asm__ __volatile__(
+            "ldrex %0, [%2]\nteq   %0, #0\nstrexeq %0, %1, [%2]"
+            : "=&r"(result)
+            : "r"(1), "r"(lock)
+            : "cc", "memory");
+        return result == 0;
+    }
+#endif
+
+    __asm__ __volatile__(
+        "swp %0, %1, [%2]\n"
+        : "=&r,&r"(result)
+        : "r,0"(1), "r,r"(lock)
+        : "memory");
+    return result == 0;
+
+#elif defined(__GNUC__) && defined(__arm__)
+    int result;
+    __asm__ __volatile__(
+        "ldrex %0, [%2]\nteq   %0, #0\nstrexeq %0, %1, [%2]"
+        : "=&r"(result)
+        : "r"(1), "r"(lock)
+        : "cc", "memory");
+    return result == 0;
+
+#elif defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__))
+    int result;
+    __asm__ __volatile__(
+        "lock ; xchgl %0, (%1)\n"
+        : "=r"(result)
+        : "r"(lock), "0"(1)
+        : "cc", "memory");
+    return result == 0;
+
+#elif defined(__MACOSX__) || defined(__IPHONEOS__)
+    /* Maybe used for PowerPC, but the Intel asm or gcc atomics are favored. */
+    return OSAtomicCompareAndSwap32Barrier(0, 1, lock);
+
+#elif defined(__SOLARIS__) && defined(_LP64)
+    /* Used for Solaris with non-gcc compilers. */
+    return (SDL_bool)((int)atomic_cas_64((volatile uint64_t *)lock, 0, 1) == 0);
+
+#elif defined(__SOLARIS__) && !defined(_LP64)
+    /* Used for Solaris with non-gcc compilers. */
+    return (SDL_bool)((int)atomic_cas_32((volatile uint32_t *)lock, 0, 1) == 0);
+#elif defined(PS2)
+    uint32_t oldintr;
+    SDL_bool res = SDL_FALSE;
+    // disable interuption
+    oldintr = DIntr();
+
+    if (*lock == 0) {
+        *lock = 1;
+        res = SDL_TRUE;
+    }
+    // enable interuption
+    if (oldintr) {
+        EIntr();
+    }
+    return res;
+#else
+#error Please implement for your platform.
+    return SDL_FALSE;
+#endif
+}
+
+void SDL_AtomicLock(SDL_SpinLock *lock)
+{
+    int iterations = 0;
+    /* FIXME: Should we have an eventual timeout? */
+    while (!SDL_AtomicTryLock(lock)) {
+        if (iterations < 32) {
+            iterations++;
+            SDL_CPUPauseInstruction();
+        } else {
+            /* !!! FIXME: this doesn't definitely give up the current timeslice, it does different things on various platforms. */
+            SDL_Delay(0);
+        }
+    }
+}
+
+void SDL_AtomicUnlock(SDL_SpinLock *lock)
+{
+#if HAVE_GCC_ATOMICS || HAVE_GCC_SYNC_LOCK_TEST_AND_SET
+    __sync_lock_release(lock);
+
+#elif defined(_MSC_VER) && (defined(_M_ARM) || defined(_M_ARM64))
+    _InterlockedExchange_rel(lock, 0);
+
+#elif defined(_MSC_VER)
+    _ReadWriteBarrier();
+    *lock = 0;
+
+#elif defined(__WATCOMC__) && defined(__386__)
+    SDL_CompilerBarrier();
+    *lock = 0;
+
+#elif defined(__SOLARIS__)
+    /* Used for Solaris when not using gcc. */
+    *lock = 0;
+    membar_producer();
+
+#else
+    *lock = 0;
+#endif
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/SDL_audio_c.h

@@ -0,0 +1,76 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_audio_c_h_
+#define SDL_audio_c_h_
+
+#include "../SDL_internal.h"
+
+#ifndef DEBUG_CONVERT
+#define DEBUG_CONVERT 0
+#endif
+
+#if DEBUG_CONVERT
+#define LOG_DEBUG_CONVERT(from, to) SDL_Log("SDL_AUDIO_CONVERT: Converting %s to %s.\n", from, to);
+#else
+#define LOG_DEBUG_CONVERT(from, to)
+#endif
+
+/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
+
+#ifdef HAVE_LIBSAMPLERATE_H
+#include "samplerate.h"
+extern SDL_bool SRC_available;
+extern int SRC_converter;
+extern SRC_STATE *(*SRC_src_new)(int converter_type, int channels, int *error);
+extern int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data);
+extern int (*SRC_src_reset)(SRC_STATE *state);
+extern SRC_STATE *(*SRC_src_delete)(SRC_STATE *state);
+extern const char *(*SRC_src_strerror)(int error);
+extern int (*SRC_src_simple)(SRC_DATA *data, int converter_type, int channels);
+#endif
+
+/* Functions to get a list of "close" audio formats */
+extern SDL_AudioFormat SDL_FirstAudioFormat(SDL_AudioFormat format);
+extern SDL_AudioFormat SDL_NextAudioFormat(void);
+
+/* Function to calculate the size and silence for a SDL_AudioSpec */
+extern Uint8 SDL_SilenceValueForFormat(const SDL_AudioFormat format);
+extern void SDL_CalculateAudioSpec(SDL_AudioSpec *spec);
+
+/* Choose the audio filter functions below */
+extern void SDL_ChooseAudioConverters(void);
+
+/* These pointers get set during SDL_ChooseAudioConverters() to various SIMD implementations. */
+extern SDL_AudioFilter SDL_Convert_S8_to_F32;
+extern SDL_AudioFilter SDL_Convert_U8_to_F32;
+extern SDL_AudioFilter SDL_Convert_S16_to_F32;
+extern SDL_AudioFilter SDL_Convert_U16_to_F32;
+extern SDL_AudioFilter SDL_Convert_S32_to_F32;
+extern SDL_AudioFilter SDL_Convert_F32_to_S8;
+extern SDL_AudioFilter SDL_Convert_F32_to_U8;
+extern SDL_AudioFilter SDL_Convert_F32_to_S16;
+extern SDL_AudioFilter SDL_Convert_F32_to_U16;
+extern SDL_AudioFilter SDL_Convert_F32_to_S32;
+
+#endif /* SDL_audio_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/SDL_audiodev.c

@@ -0,0 +1,128 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+/* Get the name of the audio device we use for output */
+
+#if SDL_AUDIO_DRIVER_NETBSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO
+
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h> /* For close() */
+
+#include "SDL_stdinc.h"
+#include "SDL_audiodev_c.h"
+
+#ifndef _PATH_DEV_DSP
+#if defined(__NETBSD__) || defined(__OPENBSD__)
+#define _PATH_DEV_DSP "/dev/audio"
+#else
+#define _PATH_DEV_DSP "/dev/dsp"
+#endif
+#endif
+#ifndef _PATH_DEV_DSP24
+#define _PATH_DEV_DSP24 "/dev/sound/dsp"
+#endif
+#ifndef _PATH_DEV_AUDIO
+#define _PATH_DEV_AUDIO "/dev/audio"
+#endif
+
+static void test_device(const int iscapture, const char *fname, int flags, int (*test)(int fd))
+{
+    struct stat sb;
+    if ((stat(fname, &sb) == 0) && (S_ISCHR(sb.st_mode))) {
+        const int audio_fd = open(fname, flags | O_CLOEXEC, 0);
+        if (audio_fd >= 0) {
+            const int okay = test(audio_fd);
+            close(audio_fd);
+            if (okay) {
+                static size_t dummyhandle = 0;
+                dummyhandle++;
+                SDL_assert(dummyhandle != 0);
+
+                /* Note that spec is NULL; while we are opening the device
+                 * endpoint here, the endpoint does not provide any mix format
+                 * information,  making this information inaccessible at
+                 * enumeration time
+                 */
+                SDL_AddAudioDevice(iscapture, fname, NULL, (void *)(uintptr_t)dummyhandle);
+            }
+        }
+    }
+}
+
+static int test_stub(int fd)
+{
+    return 1;
+}
+
+static void SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*test)(int))
+{
+    const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
+    const char *audiodev;
+    char audiopath[1024];
+
+    if (test == NULL) {
+        test = test_stub;
+    }
+
+    /* Figure out what our audio device is */
+    audiodev = SDL_getenv("SDL_PATH_DSP");
+    if (audiodev == NULL) {
+        audiodev = SDL_getenv("AUDIODEV");
+    }
+    if (audiodev == NULL) {
+        if (classic) {
+            audiodev = _PATH_DEV_AUDIO;
+        } else {
+            struct stat sb;
+
+            /* Added support for /dev/sound/\* in Linux 2.4 */
+            if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode)) && ((stat(_PATH_DEV_DSP24, &sb) == 0) && S_ISCHR(sb.st_mode))) {
+                audiodev = _PATH_DEV_DSP24;
+            } else {
+                audiodev = _PATH_DEV_DSP;
+            }
+        }
+    }
+    test_device(iscapture, audiodev, flags, test);
+
+    if (SDL_strlen(audiodev) < (sizeof(audiopath) - 3)) {
+        int instance = 0;
+        while (instance <= 64) {
+            (void)SDL_snprintf(audiopath, SDL_arraysize(audiopath),
+                               "%s%d", audiodev, instance);
+            instance++;
+            test_device(iscapture, audiopath, flags, test);
+        }
+    }
+}
+
+void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int))
+{
+    SDL_EnumUnixAudioDevices_Internal(SDL_TRUE, classic, test);
+    SDL_EnumUnixAudioDevices_Internal(SDL_FALSE, classic, test);
+}
+
+#endif /* Audio driver selection */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/SDL_audiodev_c.h

@@ -0,0 +1,44 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_audiodev_c_h_
+#define SDL_audiodev_c_h_
+
+#include "SDL.h"
+#include "../SDL_internal.h"
+#include "SDL_sysaudio.h"
+
+/* Open the audio device for playback, and don't block if busy */
+/* #define USE_BLOCKING_WRITES */
+
+#ifdef USE_BLOCKING_WRITES
+#define OPEN_FLAGS_OUTPUT O_WRONLY
+#define OPEN_FLAGS_INPUT  O_RDONLY
+#else
+#define OPEN_FLAGS_OUTPUT (O_WRONLY | O_NONBLOCK)
+#define OPEN_FLAGS_INPUT  (O_RDONLY | O_NONBLOCK)
+#endif
+
+extern void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int));
+
+#endif /* SDL_audiodev_c_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/SDL_mixer.c

@@ -0,0 +1,345 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+/* This provides the default mixing callback for the SDL audio routines */
+
+#include "SDL_cpuinfo.h"
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "SDL_sysaudio.h"
+
+/* This table is used to add two sound values together and pin
+ * the value to avoid overflow.  (used with permission from ARDI)
+ */
+static const Uint8 mix8[] = {
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
+    0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
+    0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
+    0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
+    0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
+    0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
+    0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
+    0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
+    0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
+    0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
+    0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
+    0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
+    0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
+    0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
+    0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
+    0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
+    0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
+    0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
+    0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
+    0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
+    0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
+    0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
+    0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
+    0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF
+};
+
+/* The volume ranges from 0 - 128 */
+#define ADJUST_VOLUME(s, v)     (s = (s * v) / SDL_MIX_MAXVOLUME)
+#define ADJUST_VOLUME_U8(s, v)  (s = (((s - 128) * v) / SDL_MIX_MAXVOLUME) + 128)
+#define ADJUST_VOLUME_U16(s, v) (s = (((s - 32768) * v) / SDL_MIX_MAXVOLUME) + 32768)
+
+void SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
+                        Uint32 len, int volume)
+{
+    if (volume == 0) {
+        return;
+    }
+
+    switch (format) {
+
+    case AUDIO_U8:
+    {
+        Uint8 src_sample;
+
+        while (len--) {
+            src_sample = *src;
+            ADJUST_VOLUME_U8(src_sample, volume);
+            *dst = mix8[*dst + src_sample];
+            ++dst;
+            ++src;
+        }
+    } break;
+
+    case AUDIO_S8:
+    {
+        Sint8 *dst8, *src8;
+        Sint8 src_sample;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT8;
+        const int min_audioval = SDL_MIN_SINT8;
+
+        src8 = (Sint8 *)src;
+        dst8 = (Sint8 *)dst;
+        while (len--) {
+            src_sample = *src8;
+            ADJUST_VOLUME(src_sample, volume);
+            dst_sample = *dst8 + src_sample;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *dst8 = dst_sample;
+            ++dst8;
+            ++src8;
+        }
+    } break;
+
+    case AUDIO_S16LSB:
+    {
+        Sint16 src1, src2;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT16;
+        const int min_audioval = SDL_MIN_SINT16;
+
+        len /= 2;
+        while (len--) {
+            src1 = SDL_SwapLE16(*(Sint16 *)src);
+            ADJUST_VOLUME(src1, volume);
+            src2 = SDL_SwapLE16(*(Sint16 *)dst);
+            src += 2;
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(Sint16 *)dst = SDL_SwapLE16(dst_sample);
+            dst += 2;
+        }
+    } break;
+
+    case AUDIO_S16MSB:
+    {
+        Sint16 src1, src2;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT16;
+        const int min_audioval = SDL_MIN_SINT16;
+
+        len /= 2;
+        while (len--) {
+            src1 = SDL_SwapBE16(*(Sint16 *)src);
+            ADJUST_VOLUME(src1, volume);
+            src2 = SDL_SwapBE16(*(Sint16 *)dst);
+            src += 2;
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(Sint16 *)dst = SDL_SwapBE16(dst_sample);
+            dst += 2;
+        }
+    } break;
+
+    case AUDIO_U16LSB:
+    {
+        Uint16 src1, src2;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT16;
+        const int min_audioval = SDL_MIN_SINT16;
+
+        len /= 2;
+        while (len--) {
+            src1 = SDL_SwapLE16(*(Uint16 *)src);
+            ADJUST_VOLUME_U16(src1, volume);
+            src2 = SDL_SwapLE16(*(Uint16 *)dst);
+            src += 2;
+            dst_sample = src1 + src2 - 32768 * 2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            dst_sample += 32768;
+            *(Uint16 *)dst = SDL_SwapLE16(dst_sample);
+            dst += 2;
+        }
+    } break;
+
+    case AUDIO_U16MSB:
+    {
+        Uint16 src1, src2;
+        int dst_sample;
+        const int max_audioval = SDL_MAX_SINT16;
+        const int min_audioval = SDL_MIN_SINT16;
+
+        len /= 2;
+        while (len--) {
+            src1 = SDL_SwapBE16(*(Uint16 *)src);
+            ADJUST_VOLUME_U16(src1, volume);
+            src2 = SDL_SwapBE16(*(Uint16 *)dst);
+            src += 2;
+            dst_sample = src1 + src2 - 32768 * 2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            dst_sample += 32768;
+            *(Uint16 *)dst = SDL_SwapBE16(dst_sample);
+            dst += 2;
+        }
+    } break;
+
+    case AUDIO_S32LSB:
+    {
+        const Uint32 *src32 = (Uint32 *)src;
+        Uint32 *dst32 = (Uint32 *)dst;
+        Sint64 src1, src2;
+        Sint64 dst_sample;
+        const Sint64 max_audioval = SDL_MAX_SINT32;
+        const Sint64 min_audioval = SDL_MIN_SINT32;
+
+        len /= 4;
+        while (len--) {
+            src1 = (Sint64)((Sint32)SDL_SwapLE32(*src32));
+            src32++;
+            ADJUST_VOLUME(src1, volume);
+            src2 = (Sint64)((Sint32)SDL_SwapLE32(*dst32));
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_SwapLE32((Uint32)((Sint32)dst_sample));
+        }
+    } break;
+
+    case AUDIO_S32MSB:
+    {
+        const Uint32 *src32 = (Uint32 *)src;
+        Uint32 *dst32 = (Uint32 *)dst;
+        Sint64 src1, src2;
+        Sint64 dst_sample;
+        const Sint64 max_audioval = SDL_MAX_SINT32;
+        const Sint64 min_audioval = SDL_MIN_SINT32;
+
+        len /= 4;
+        while (len--) {
+            src1 = (Sint64)((Sint32)SDL_SwapBE32(*src32));
+            src32++;
+            ADJUST_VOLUME(src1, volume);
+            src2 = (Sint64)((Sint32)SDL_SwapBE32(*dst32));
+            dst_sample = src1 + src2;
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_SwapBE32((Uint32)((Sint32)dst_sample));
+        }
+    } break;
+
+    case AUDIO_F32LSB:
+    {
+        const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME);
+        const float fvolume = (float)volume;
+        const float *src32 = (float *)src;
+        float *dst32 = (float *)dst;
+        float src1, src2;
+        double dst_sample;
+        /* !!! FIXME: are these right? */
+        const double max_audioval = 3.402823466e+38F;
+        const double min_audioval = -3.402823466e+38F;
+
+        len /= 4;
+        while (len--) {
+            src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
+            src2 = SDL_SwapFloatLE(*dst32);
+            src32++;
+
+            dst_sample = ((double)src1) + ((double)src2);
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_SwapFloatLE((float)dst_sample);
+        }
+    } break;
+
+    case AUDIO_F32MSB:
+    {
+        const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME);
+        const float fvolume = (float)volume;
+        const float *src32 = (float *)src;
+        float *dst32 = (float *)dst;
+        float src1, src2;
+        double dst_sample;
+        /* !!! FIXME: are these right? */
+        const double max_audioval = 3.402823466e+38F;
+        const double min_audioval = -3.402823466e+38F;
+
+        len /= 4;
+        while (len--) {
+            src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
+            src2 = SDL_SwapFloatBE(*dst32);
+            src32++;
+
+            dst_sample = ((double)src1) + ((double)src2);
+            if (dst_sample > max_audioval) {
+                dst_sample = max_audioval;
+            } else if (dst_sample < min_audioval) {
+                dst_sample = min_audioval;
+            }
+            *(dst32++) = SDL_SwapFloatBE((float)dst_sample);
+        }
+    } break;
+
+    default: /* If this happens... FIXME! */
+        SDL_SetError("SDL_MixAudioFormat(): unknown audio format");
+        return;
+    }
+}
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/SDL_sysaudio.h

@@ -0,0 +1,215 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+#ifndef SDL_sysaudio_h_
+#define SDL_sysaudio_h_
+
+#include "SDL_mutex.h"
+#include "SDL_thread.h"
+#include "../SDL_dataqueue.h"
+#include "./SDL_audio_c.h"
+
+/* !!! FIXME: These are wordy and unlocalized... */
+#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
+#define DEFAULT_INPUT_DEVNAME  "System audio capture device"
+
+/* The SDL audio driver */
+typedef struct SDL_AudioDevice SDL_AudioDevice;
+#define _THIS SDL_AudioDevice *_this
+
+/* Audio targets should call this as devices are added to the system (such as
+   a USB headset being plugged in), and should also be called for
+   for every device found during DetectDevices(). */
+extern void SDL_AddAudioDevice(const SDL_bool iscapture, const char *name, SDL_AudioSpec *spec, void *handle);
+
+/* Audio targets should call this as devices are removed, so SDL can update
+   its list of available devices. */
+extern void SDL_RemoveAudioDevice(const SDL_bool iscapture, void *handle);
+
+/* Audio targets should call this if an opened audio device is lost while
+   being used. This can happen due to i/o errors, or a device being unplugged,
+   etc. If the device is totally gone, please also call SDL_RemoveAudioDevice()
+   as appropriate so SDL's list of devices is accurate. */
+extern void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device);
+
+/* This is the size of a packet when using SDL_QueueAudio(). We allocate
+   these as necessary and pool them, under the assumption that we'll
+   eventually end up with a handful that keep recycling, meeting whatever
+   the app needs. We keep packing data tightly as more arrives to avoid
+   wasting space, and if we get a giant block of data, we'll split them
+   into multiple packets behind the scenes. My expectation is that most
+   apps will have 2-3 of these in the pool. 8k should cover most needs, but
+   if this is crippling for some embedded system, we can #ifdef this.
+   The system preallocates enough packets for 2 callbacks' worth of data. */
+#define SDL_AUDIOBUFFERQUEUE_PACKETLEN (8 * 1024)
+
+typedef struct SDL_AudioDriverImpl
+{
+    void (*DetectDevices)(void);
+    int (*OpenDevice)(_THIS, const char *devname);
+    void (*ThreadInit)(_THIS);   /* Called by audio thread at start */
+    void (*ThreadDeinit)(_THIS); /* Called by audio thread at end */
+    void (*WaitDevice)(_THIS);
+    void (*PlayDevice)(_THIS);
+    Uint8 *(*GetDeviceBuf)(_THIS);
+    int (*CaptureFromDevice)(_THIS, void *buffer, int buflen);
+    void (*FlushCapture)(_THIS);
+    void (*CloseDevice)(_THIS);
+    void (*LockDevice)(_THIS);
+    void (*UnlockDevice)(_THIS);
+    void (*FreeDeviceHandle)(void *handle); /**< SDL is done with handle from SDL_AddAudioDevice() */
+    void (*Deinitialize)(void);
+    int (*GetDefaultAudioInfo)(char **name, SDL_AudioSpec *spec, int iscapture);
+
+    /* !!! FIXME: add pause(), so we can optimize instead of mixing silence. */
+
+    /* Some flags to push duplicate code into the core and reduce #ifdefs. */
+    SDL_bool ProvidesOwnCallbackThread;
+    SDL_bool HasCaptureSupport;
+    SDL_bool OnlyHasDefaultOutputDevice;
+    SDL_bool OnlyHasDefaultCaptureDevice;
+    SDL_bool AllowsArbitraryDeviceNames;
+    SDL_bool SupportsNonPow2Samples;
+} SDL_AudioDriverImpl;
+
+typedef struct SDL_AudioDeviceItem
+{
+    void *handle;
+    char *name;
+    char *original_name;
+    SDL_AudioSpec spec;
+    int dupenum;
+    struct SDL_AudioDeviceItem *next;
+} SDL_AudioDeviceItem;
+
+typedef struct SDL_AudioDriver
+{
+    /* * * */
+    /* The name of this audio driver */
+    const char *name;
+
+    /* * * */
+    /* The description of this audio driver */
+    const char *desc;
+
+    SDL_AudioDriverImpl impl;
+
+    /* A mutex for device detection */
+    SDL_mutex *detectionLock;
+    SDL_bool captureDevicesRemoved;
+    SDL_bool outputDevicesRemoved;
+    int outputDeviceCount;
+    int inputDeviceCount;
+    SDL_AudioDeviceItem *outputDevices;
+    SDL_AudioDeviceItem *inputDevices;
+} SDL_AudioDriver;
+
+/* Define the SDL audio driver structure */
+struct SDL_AudioDevice
+{
+    /* * * */
+    /* Data common to all devices */
+    SDL_AudioDeviceID id;
+
+    /* The device's current audio specification */
+    SDL_AudioSpec spec;
+
+    /* The callback's expected audio specification (converted vs device's spec). */
+    SDL_AudioSpec callbackspec;
+
+    /* Stream that converts and resamples. NULL if not needed. */
+    SDL_AudioStream *stream;
+
+    /* Current state flags */
+    SDL_atomic_t shutdown; /* true if we are signaling the play thread to end. */
+    SDL_atomic_t enabled;  /* true if device is functioning and connected. */
+    SDL_atomic_t paused;
+    SDL_bool iscapture;
+
+    /* Scratch buffer used in the bridge between SDL and the user callback. */
+    Uint8 *work_buffer;
+
+    /* Size, in bytes, of work_buffer. */
+    Uint32 work_buffer_len;
+
+    /* A mutex for locking the mixing buffers */
+    SDL_mutex *mixer_lock;
+
+    /* A thread to feed the audio device */
+    SDL_Thread *thread;
+    SDL_threadID threadid;
+
+    /* Queued buffers (if app not using callback). */
+    SDL_DataQueue *buffer_queue;
+
+    /* * * */
+    /* Data private to this driver */
+    struct SDL_PrivateAudioData *hidden;
+
+    void *handle;
+};
+#undef _THIS
+
+typedef struct AudioBootStrap
+{
+    const char *name;
+    const char *desc;
+    SDL_bool (*init)(SDL_AudioDriverImpl *impl);
+    SDL_bool demand_only; /* 1==request explicitly, or it won't be available. */
+} AudioBootStrap;
+
+/* Not all of these are available in a given build. Use #ifdefs, etc. */
+extern AudioBootStrap PIPEWIRE_bootstrap;
+extern AudioBootStrap PULSEAUDIO_bootstrap;
+extern AudioBootStrap ALSA_bootstrap;
+extern AudioBootStrap JACK_bootstrap;
+extern AudioBootStrap SNDIO_bootstrap;
+extern AudioBootStrap NETBSDAUDIO_bootstrap;
+extern AudioBootStrap DSP_bootstrap;
+extern AudioBootStrap QSAAUDIO_bootstrap;
+extern AudioBootStrap SUNAUDIO_bootstrap;
+extern AudioBootStrap ARTS_bootstrap;
+extern AudioBootStrap ESD_bootstrap;
+extern AudioBootStrap NACLAUDIO_bootstrap;
+extern AudioBootStrap NAS_bootstrap;
+extern AudioBootStrap WASAPI_bootstrap;
+extern AudioBootStrap DSOUND_bootstrap;
+extern AudioBootStrap WINMM_bootstrap;
+extern AudioBootStrap PAUDIO_bootstrap;
+extern AudioBootStrap HAIKUAUDIO_bootstrap;
+extern AudioBootStrap COREAUDIO_bootstrap;
+extern AudioBootStrap DISKAUDIO_bootstrap;
+extern AudioBootStrap DUMMYAUDIO_bootstrap;
+extern AudioBootStrap FUSIONSOUND_bootstrap;
+extern AudioBootStrap aaudio_bootstrap;
+extern AudioBootStrap openslES_bootstrap;
+extern AudioBootStrap ANDROIDAUDIO_bootstrap;
+extern AudioBootStrap PS2AUDIO_bootstrap;
+extern AudioBootStrap PSPAUDIO_bootstrap;
+extern AudioBootStrap VITAAUD_bootstrap;
+extern AudioBootStrap N3DSAUDIO_bootstrap;
+extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;
+extern AudioBootStrap OS2AUDIO_bootstrap;
+
+#endif /* SDL_sysaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/SDL_wave.h

@@ -0,0 +1,153 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+/* RIFF WAVE files are little-endian */
+
+/*******************************************/
+/* Define values for Microsoft WAVE format */
+/*******************************************/
+/* FOURCC */
+#define RIFF 0x46464952 /* "RIFF" */
+#define WAVE 0x45564157 /* "WAVE" */
+#define FACT 0x74636166 /* "fact" */
+#define LIST 0x5453494c /* "LIST" */
+#define BEXT 0x74786562 /* "bext" */
+#define JUNK 0x4B4E554A /* "JUNK" */
+#define FMT  0x20746D66 /* "fmt " */
+#define DATA 0x61746164 /* "data" */
+/* Format tags */
+#define UNKNOWN_CODE    0x0000
+#define PCM_CODE        0x0001
+#define MS_ADPCM_CODE   0x0002
+#define IEEE_FLOAT_CODE 0x0003
+#define ALAW_CODE       0x0006
+#define MULAW_CODE      0x0007
+#define IMA_ADPCM_CODE  0x0011
+#define MPEG_CODE       0x0050
+#define MPEGLAYER3_CODE 0x0055
+#define EXTENSIBLE_CODE 0xFFFE
+
+/* Stores the WAVE format information. */
+typedef struct WaveFormat
+{
+    Uint16 formattag;     /* Raw value of the first field in the fmt chunk data. */
+    Uint16 encoding;      /* Actual encoding, possibly from the extensible header. */
+    Uint16 channels;      /* Number of channels. */
+    Uint32 frequency;     /* Sampling rate in Hz. */
+    Uint32 byterate;      /* Average bytes per second. */
+    Uint16 blockalign;    /* Bytes per block. */
+    Uint16 bitspersample; /* Currently supported are 8, 16, 24, 32, and 4 for ADPCM. */
+
+    /* Extra information size. Number of extra bytes starting at byte 18 in the
+     * fmt chunk data. This is at least 22 for the extensible header.
+     */
+    Uint16 extsize;
+
+    /* Extensible WAVE header fields */
+    Uint16 validsamplebits;
+    Uint32 samplesperblock; /* For compressed formats. Can be zero. Actually 16 bits in the header. */
+    Uint32 channelmask;
+    Uint8 subformat[16]; /* A format GUID. */
+} WaveFormat;
+
+/* Stores information on the fact chunk. */
+typedef struct WaveFact
+{
+    /* Represents the state of the fact chunk in the WAVE file.
+     * Set to -1 if the fact chunk is invalid.
+     * Set to 0 if the fact chunk is not present
+     * Set to 1 if the fact chunk is present and valid.
+     * Set to 2 if samplelength is going to be used as the number of sample frames.
+     */
+    Sint32 status;
+
+    /* Version 1 of the RIFF specification calls the field in the fact chunk
+     * dwFileSize. The Standards Update then calls it dwSampleLength and specifies
+     * that it is 'the length of the data in samples'. WAVE files from Windows
+     * with this chunk have it set to the samples per channel (sample frames).
+     * This is useful to truncate compressed audio to a specific sample count
+     * because a compressed block is usually decoded to a fixed number of
+     * sample frames.
+     */
+    Uint32 samplelength; /* Raw sample length value from the fact chunk. */
+} WaveFact;
+
+/* Generic struct for the chunks in the WAVE file. */
+typedef struct WaveChunk
+{
+    Uint32 fourcc;   /* FOURCC of the chunk. */
+    Uint32 length;   /* Size of the chunk data. */
+    Sint64 position; /* Position of the data in the stream. */
+    Uint8 *data;     /* When allocated, this points to the chunk data. length is used for the memory allocation size. */
+    size_t size;     /* Number of bytes in data that could be read from the stream. Can be smaller than length. */
+} WaveChunk;
+
+/* Controls how the size of the RIFF chunk affects the loading of a WAVE file. */
+typedef enum WaveRiffSizeHint
+{
+    RiffSizeNoHint,
+    RiffSizeForce,
+    RiffSizeIgnoreZero,
+    RiffSizeIgnore,
+    RiffSizeMaximum
+} WaveRiffSizeHint;
+
+/* Controls how a truncated WAVE file is handled. */
+typedef enum WaveTruncationHint
+{
+    TruncNoHint,
+    TruncVeryStrict,
+    TruncStrict,
+    TruncDropFrame,
+    TruncDropBlock
+} WaveTruncationHint;
+
+/* Controls how the fact chunk affects the loading of a WAVE file. */
+typedef enum WaveFactChunkHint
+{
+    FactNoHint,
+    FactTruncate,
+    FactStrict,
+    FactIgnoreZero,
+    FactIgnore
+} WaveFactChunkHint;
+
+typedef struct WaveFile
+{
+    WaveChunk chunk;
+    WaveFormat format;
+    WaveFact fact;
+
+    /* Number of sample frames that will be decoded. Calculated either with the
+     * size of the data chunk or, if the appropriate hint is enabled, with the
+     * sample length value from the fact chunk.
+     */
+    Sint64 sampleframes;
+
+    void *decoderdata; /* Some decoders require extra data for a state. */
+
+    WaveRiffSizeHint riffhint;
+    WaveTruncationHint trunchint;
+    WaveFactChunkHint facthint;
+} WaveFile;
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/aaudio/SDL_aaudio.c

@@ -0,0 +1,465 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_AAUDIO
+
+#include "SDL_audio.h"
+#include "SDL_loadso.h"
+#include "../SDL_audio_c.h"
+#include "../../core/android/SDL_android.h"
+#include "SDL_aaudio.h"
+
+/* Debug */
+#if 0
+#define LOGI(...) SDL_Log(__VA_ARGS__);
+#else
+#define LOGI(...)
+#endif
+
+typedef struct AAUDIO_Data
+{
+    AAudioStreamBuilder *builder;
+    void *handle;
+#define SDL_PROC(ret, func, params) ret (*func) params;
+#include "SDL_aaudiofuncs.h"
+#undef SDL_PROC
+} AAUDIO_Data;
+static AAUDIO_Data ctx;
+
+static SDL_AudioDevice *audioDevice = NULL;
+static SDL_AudioDevice *captureDevice = NULL;
+
+static int aaudio_LoadFunctions(AAUDIO_Data *data)
+{
+#define SDL_PROC(ret, func, params)                                                             \
+    do {                                                                                        \
+        data->func = SDL_LoadFunction(data->handle, #func);                                     \
+        if (!data->func) {                                                                      \
+            return SDL_SetError("Couldn't load AAUDIO function %s: %s", #func, SDL_GetError()); \
+        }                                                                                       \
+    } while (0);
+#include "SDL_aaudiofuncs.h"
+#undef SDL_PROC
+    return 0;
+}
+
+void aaudio_errorCallback(AAudioStream *stream, void *userData, aaudio_result_t error);
+void aaudio_errorCallback(AAudioStream *stream, void *userData, aaudio_result_t error)
+{
+    LOGI("SDL aaudio_errorCallback: %d - %s", error, ctx.AAudio_convertResultToText(error));
+}
+
+#define LIB_AAUDIO_SO "libaaudio.so"
+
+static int aaudio_OpenDevice(_THIS, const char *devname)
+{
+    struct SDL_PrivateAudioData *private;
+    SDL_bool iscapture = this->iscapture;
+    aaudio_result_t res;
+    LOGI(__func__);
+
+    SDL_assert((captureDevice == NULL) || !iscapture);
+    SDL_assert((audioDevice == NULL) || iscapture);
+
+    if (iscapture) {
+        if (!Android_JNI_RequestPermission("android.permission.RECORD_AUDIO")) {
+            LOGI("This app doesn't have RECORD_AUDIO permission");
+            return SDL_SetError("This app doesn't have RECORD_AUDIO permission");
+        }
+    }
+
+    if (iscapture) {
+        captureDevice = this;
+    } else {
+        audioDevice = this;
+    }
+
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    private = this->hidden;
+
+    ctx.AAudioStreamBuilder_setSampleRate(ctx.builder, this->spec.freq);
+    ctx.AAudioStreamBuilder_setChannelCount(ctx.builder, this->spec.channels);
+    if(devname != NULL) {
+        int aaudio_device_id = SDL_atoi(devname);
+        LOGI("Opening device id %d", aaudio_device_id);
+        ctx.AAudioStreamBuilder_setDeviceId(ctx.builder, aaudio_device_id);
+    }
+    {
+        aaudio_direction_t direction = (iscapture ? AAUDIO_DIRECTION_INPUT : AAUDIO_DIRECTION_OUTPUT);
+        ctx.AAudioStreamBuilder_setDirection(ctx.builder, direction);
+    }
+    {
+        aaudio_format_t format = AAUDIO_FORMAT_PCM_FLOAT;
+        if (this->spec.format == AUDIO_S16SYS) {
+            format = AAUDIO_FORMAT_PCM_I16;
+        } else if (this->spec.format == AUDIO_S16SYS) {
+            format = AAUDIO_FORMAT_PCM_FLOAT;
+        }
+        ctx.AAudioStreamBuilder_setFormat(ctx.builder, format);
+    }
+
+    ctx.AAudioStreamBuilder_setErrorCallback(ctx.builder, aaudio_errorCallback, private);
+
+    LOGI("AAudio Try to open %u hz %u bit chan %u %s samples %u",
+         this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
+         this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
+
+    res = ctx.AAudioStreamBuilder_openStream(ctx.builder, &private->stream);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudioStreamBuilder_openStream %d", res);
+        return SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    }
+
+    this->spec.freq = ctx.AAudioStream_getSampleRate(private->stream);
+    this->spec.channels = ctx.AAudioStream_getChannelCount(private->stream);
+    {
+        aaudio_format_t fmt = ctx.AAudioStream_getFormat(private->stream);
+        if (fmt == AAUDIO_FORMAT_PCM_I16) {
+            this->spec.format = AUDIO_S16SYS;
+        } else if (fmt == AAUDIO_FORMAT_PCM_FLOAT) {
+            this->spec.format = AUDIO_F32SYS;
+        }
+    }
+
+    LOGI("AAudio Try to open %u hz %u bit chan %u %s samples %u",
+         this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
+         this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    if (!iscapture) {
+        private->mixlen = this->spec.size;
+        private->mixbuf = (Uint8 *)SDL_malloc(private->mixlen);
+        if (private->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(private->mixbuf, this->spec.silence, this->spec.size);
+    }
+
+    private->frame_size = this->spec.channels * (SDL_AUDIO_BITSIZE(this->spec.format) / 8);
+
+    res = ctx.AAudioStream_requestStart(private->stream);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudioStream_requestStart %d iscapture:%d", res, iscapture);
+        return SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    }
+
+    LOGI("SDL AAudioStream_requestStart OK");
+    return 0;
+}
+
+static void aaudio_CloseDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *private = this->hidden;
+    aaudio_result_t res;
+    LOGI(__func__);
+
+    if (private->stream) {
+        res = ctx.AAudioStream_requestStop(private->stream);
+        if (res != AAUDIO_OK) {
+            LOGI("SDL Failed AAudioStream_requestStop %d", res);
+            SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            return;
+        }
+
+        res = ctx.AAudioStream_close(private->stream);
+        if (res != AAUDIO_OK) {
+            LOGI("SDL Failed AAudioStreamBuilder_delete %d", res);
+            SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            return;
+        }
+    }
+
+    if (this->iscapture) {
+        SDL_assert(captureDevice == this);
+        captureDevice = NULL;
+    } else {
+        SDL_assert(audioDevice == this);
+        audioDevice = NULL;
+    }
+
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static Uint8 *aaudio_GetDeviceBuf(_THIS)
+{
+    struct SDL_PrivateAudioData *private = this->hidden;
+    return private->mixbuf;
+}
+
+static void aaudio_PlayDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *private = this->hidden;
+    aaudio_result_t res;
+    int64_t timeoutNanoseconds = 1 * 1000 * 1000; /* 8 ms */
+    res = ctx.AAudioStream_write(private->stream, private->mixbuf, private->mixlen / private->frame_size, timeoutNanoseconds);
+    if (res < 0) {
+        LOGI("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    } else {
+        LOGI("SDL AAudio play: %d frames, wanted:%d frames", (int)res, private->mixlen / private->frame_size);
+    }
+
+#if 0
+    /* Log under-run count */
+    {
+        static int prev = 0;
+        int32_t cnt = ctx.AAudioStream_getXRunCount(private->stream);
+        if (cnt != prev) {
+            SDL_Log("AAudio underrun: %d - total: %d", cnt - prev, cnt);
+            prev = cnt;
+        }
+    }
+#endif
+}
+
+static int aaudio_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *private = this->hidden;
+    aaudio_result_t res;
+    int64_t timeoutNanoseconds = 8 * 1000 * 1000; /* 8 ms */
+    res = ctx.AAudioStream_read(private->stream, buffer, buflen / private->frame_size, timeoutNanoseconds);
+    if (res < 0) {
+        LOGI("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+        return -1;
+    }
+    LOGI("SDL AAudio capture:%d frames, wanted:%d frames", (int)res, buflen / private->frame_size);
+    return res * private->frame_size;
+}
+
+static void aaudio_Deinitialize(void)
+{
+    LOGI(__func__);
+    if (ctx.handle) {
+        if (ctx.builder) {
+            aaudio_result_t res;
+            res = ctx.AAudioStreamBuilder_delete(ctx.builder);
+            if (res != AAUDIO_OK) {
+                SDL_SetError("Failed AAudioStreamBuilder_delete %s", ctx.AAudio_convertResultToText(res));
+            }
+        }
+        SDL_UnloadObject(ctx.handle);
+    }
+    ctx.handle = NULL;
+    ctx.builder = NULL;
+    LOGI("End AAUDIO %s", SDL_GetError());
+}
+
+static SDL_bool aaudio_Init(SDL_AudioDriverImpl *impl)
+{
+    aaudio_result_t res;
+    LOGI(__func__);
+
+    /* AAudio was introduced in Android 8.0, but has reference counting crash issues in that release,
+     * so don't use it until 8.1.
+     *
+     * See https://github.com/google/oboe/issues/40 for more information.
+     */
+    if (SDL_GetAndroidSDKVersion() < 27) {
+        return SDL_FALSE;
+    }
+
+    SDL_zero(ctx);
+
+    ctx.handle = SDL_LoadObject(LIB_AAUDIO_SO);
+    if (ctx.handle == NULL) {
+        LOGI("SDL couldn't find " LIB_AAUDIO_SO);
+        goto failure;
+    }
+
+    if (aaudio_LoadFunctions(&ctx) < 0) {
+        goto failure;
+    }
+
+    res = ctx.AAudio_createStreamBuilder(&ctx.builder);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudio_createStreamBuilder %d", res);
+        goto failure;
+    }
+
+    if (ctx.builder == NULL) {
+        LOGI("SDL Failed AAudio_createStreamBuilder - builder NULL");
+        goto failure;
+    }
+
+    impl->DetectDevices = Android_DetectDevices;
+    impl->Deinitialize = aaudio_Deinitialize;
+    impl->OpenDevice = aaudio_OpenDevice;
+    impl->CloseDevice = aaudio_CloseDevice;
+    impl->PlayDevice = aaudio_PlayDevice;
+    impl->GetDeviceBuf = aaudio_GetDeviceBuf;
+    impl->CaptureFromDevice = aaudio_CaptureFromDevice;
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+
+    /* and the capabilities */
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->OnlyHasDefaultOutputDevice = SDL_FALSE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_FALSE;
+
+    /* this audio target is available. */
+    LOGI("SDL aaudio_Init OK");
+    return SDL_TRUE;
+
+failure:
+    if (ctx.handle) {
+        if (ctx.builder) {
+            ctx.AAudioStreamBuilder_delete(ctx.builder);
+        }
+        SDL_UnloadObject(ctx.handle);
+    }
+    ctx.handle = NULL;
+    ctx.builder = NULL;
+    return SDL_FALSE;
+}
+
+AudioBootStrap aaudio_bootstrap = {
+    "AAudio", "AAudio audio driver", aaudio_Init, SDL_FALSE
+};
+
+/* Pause (block) all non already paused audio devices by taking their mixer lock */
+void aaudio_PauseDevices(void)
+{
+    /* TODO: Handle multiple devices? */
+    struct SDL_PrivateAudioData *private;
+    if (audioDevice != NULL && audioDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
+
+        if (private->stream) {
+            aaudio_result_t res = ctx.AAudioStream_requestPause(private->stream);
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestPause %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+
+        if (SDL_AtomicGet(&audioDevice->paused)) {
+            /* The device is already paused, leave it alone */
+            private->resume = SDL_FALSE;
+        } else {
+            SDL_LockMutex(audioDevice->mixer_lock);
+            SDL_AtomicSet(&audioDevice->paused, 1);
+            private->resume = SDL_TRUE;
+        }
+    }
+
+    if (captureDevice != NULL && captureDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
+
+        if (private->stream) {
+            /* Pause() isn't implemented for 'capture', use Stop() */
+            aaudio_result_t res = ctx.AAudioStream_requestStop(private->stream);
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestStop %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+
+        if (SDL_AtomicGet(&captureDevice->paused)) {
+            /* The device is already paused, leave it alone */
+            private->resume = SDL_FALSE;
+        } else {
+            SDL_LockMutex(captureDevice->mixer_lock);
+            SDL_AtomicSet(&captureDevice->paused, 1);
+            private->resume = SDL_TRUE;
+        }
+    }
+}
+
+/* Resume (unblock) all non already paused audio devices by releasing their mixer lock */
+void aaudio_ResumeDevices(void)
+{
+    /* TODO: Handle multiple devices? */
+    struct SDL_PrivateAudioData *private;
+    if (audioDevice != NULL && audioDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
+
+        if (private->resume) {
+            SDL_AtomicSet(&audioDevice->paused, 0);
+            private->resume = SDL_FALSE;
+            SDL_UnlockMutex(audioDevice->mixer_lock);
+        }
+
+        if (private->stream) {
+            aaudio_result_t res = ctx.AAudioStream_requestStart(private->stream);
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestStart %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+    }
+
+    if (captureDevice != NULL && captureDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
+
+        if (private->resume) {
+            SDL_AtomicSet(&captureDevice->paused, 0);
+            private->resume = SDL_FALSE;
+            SDL_UnlockMutex(captureDevice->mixer_lock);
+        }
+
+        if (private->stream) {
+            aaudio_result_t res = ctx.AAudioStream_requestStart(private->stream);
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestStart %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+    }
+}
+
+/*
+ We can sometimes get into a state where AAudioStream_write() will just block forever until we pause and unpause.
+ None of the standard state queries indicate any problem in my testing. And the error callback doesn't actually get called.
+ But, AAudioStream_getTimestamp() does return AAUDIO_ERROR_INVALID_STATE
+*/
+SDL_bool aaudio_DetectBrokenPlayState(void)
+{
+    struct SDL_PrivateAudioData *private;
+    int64_t framePosition, timeNanoseconds;
+    aaudio_result_t res;
+
+    if (audioDevice == NULL || !audioDevice->hidden) {
+        return SDL_FALSE;
+    }
+
+    private = audioDevice->hidden;
+
+    res = ctx.AAudioStream_getTimestamp(private->stream, CLOCK_MONOTONIC, &framePosition, &timeNanoseconds);
+    if (res == AAUDIO_ERROR_INVALID_STATE) {
+        aaudio_stream_state_t currentState = ctx.AAudioStream_getState(private->stream);
+        /* AAudioStream_getTimestamp() will also return AAUDIO_ERROR_INVALID_STATE while the stream is still initially starting. But we only care if it silently went invalid while playing. */
+        if (currentState == AAUDIO_STREAM_STATE_STARTED) {
+            LOGI("SDL aaudio_DetectBrokenPlayState: detected invalid audio device state: AAudioStream_getTimestamp result=%d, framePosition=%lld, timeNanoseconds=%lld, getState=%d", (int)res, (long long)framePosition, (long long)timeNanoseconds, (int)currentState);
+            return SDL_TRUE;
+        }
+    }
+
+    return SDL_FALSE;
+}
+
+#endif /* SDL_AUDIO_DRIVER_AAUDIO */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/aaudio/SDL_aaudio.h

@@ -0,0 +1,52 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef _SDL_aaudio_h
+#define _SDL_aaudio_h
+
+#include "../SDL_sysaudio.h"
+#include <stdbool.h>
+#include <aaudio/AAudio.h>
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    AAudioStream *stream;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+    int frame_size;
+
+    /* Resume device if it was paused automatically */
+    int resume;
+};
+
+void aaudio_ResumeDevices(void);
+void aaudio_PauseDevices(void);
+SDL_bool aaudio_DetectBrokenPlayState(void);
+
+#endif /* _SDL_aaudio_h */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/aaudio/SDL_aaudiofuncs.h

@@ -0,0 +1,79 @@
+/*
+  Simple DirectMedia Layer
+  Copyright , (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#define SDL_PROC_UNUSED(ret, func, params)
+
+SDL_PROC(const char *, AAudio_convertResultToText, (aaudio_result_t returnCode))
+SDL_PROC(const char *, AAudio_convertStreamStateToText, (aaudio_stream_state_t state))
+SDL_PROC(aaudio_result_t, AAudio_createStreamBuilder, (AAudioStreamBuilder * *builder))
+SDL_PROC(void, AAudioStreamBuilder_setDeviceId, (AAudioStreamBuilder * builder, int32_t deviceId))
+SDL_PROC(void, AAudioStreamBuilder_setSampleRate, (AAudioStreamBuilder * builder, int32_t sampleRate))
+SDL_PROC(void, AAudioStreamBuilder_setChannelCount, (AAudioStreamBuilder * builder, int32_t channelCount))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSamplesPerFrame, (AAudioStreamBuilder * builder, int32_t samplesPerFrame))
+SDL_PROC(void, AAudioStreamBuilder_setFormat, (AAudioStreamBuilder * builder, aaudio_format_t format))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSharingMode, (AAudioStreamBuilder * builder, aaudio_sharing_mode_t sharingMode))
+SDL_PROC(void, AAudioStreamBuilder_setDirection, (AAudioStreamBuilder * builder, aaudio_direction_t direction))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setBufferCapacityInFrames, (AAudioStreamBuilder * builder, int32_t numFrames))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPerformanceMode, (AAudioStreamBuilder * builder, aaudio_performance_mode_t mode))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setUsage, (AAudioStreamBuilder * builder, aaudio_usage_t usage))                                         /* API 28 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setContentType, (AAudioStreamBuilder * builder, aaudio_content_type_t contentType))                      /* API 28 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setInputPreset, (AAudioStreamBuilder * builder, aaudio_input_preset_t inputPreset))                      /* API 28 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setAllowedCapturePolicy, (AAudioStreamBuilder * builder, aaudio_allowed_capture_policy_t capturePolicy)) /* API 29 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSessionId, (AAudioStreamBuilder * builder, aaudio_session_id_t sessionId))                            /* API 28 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPrivacySensitive, (AAudioStreamBuilder * builder, bool privacySensitive))                             /* API 30 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setDataCallback, (AAudioStreamBuilder * builder, AAudioStream_dataCallback callback, void *userData))
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setFramesPerDataCallback, (AAudioStreamBuilder * builder, int32_t numFrames))
+SDL_PROC(void, AAudioStreamBuilder_setErrorCallback, (AAudioStreamBuilder * builder, AAudioStream_errorCallback callback, void *userData))
+SDL_PROC(aaudio_result_t, AAudioStreamBuilder_openStream, (AAudioStreamBuilder * builder, AAudioStream **stream))
+SDL_PROC(aaudio_result_t, AAudioStreamBuilder_delete, (AAudioStreamBuilder * builder))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_release, (AAudioStream * stream)) /* API 30 */
+SDL_PROC(aaudio_result_t, AAudioStream_close, (AAudioStream * stream))
+SDL_PROC(aaudio_result_t, AAudioStream_requestStart, (AAudioStream * stream))
+SDL_PROC(aaudio_result_t, AAudioStream_requestPause, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_requestFlush, (AAudioStream * stream))
+SDL_PROC(aaudio_result_t, AAudioStream_requestStop, (AAudioStream * stream))
+SDL_PROC(aaudio_stream_state_t, AAudioStream_getState, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_waitForStateChange, (AAudioStream * stream, aaudio_stream_state_t inputState, aaudio_stream_state_t *nextState, int64_t timeoutNanoseconds))
+SDL_PROC(aaudio_result_t, AAudioStream_read, (AAudioStream * stream, void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
+SDL_PROC(aaudio_result_t, AAudioStream_write, (AAudioStream * stream, const void *buffer, int32_t numFrames, int64_t timeoutNanoseconds))
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_setBufferSizeInFrames, (AAudioStream * stream, int32_t numFrames))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferSizeInFrames, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerBurst, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getBufferCapacityInFrames, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getFramesPerDataCallback, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getXRunCount, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getSampleRate, (AAudioStream * stream))
+SDL_PROC(int32_t, AAudioStream_getChannelCount, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getSamplesPerFrame, (AAudioStream * stream))
+SDL_PROC_UNUSED(int32_t, AAudioStream_getDeviceId, (AAudioStream * stream))
+SDL_PROC(aaudio_format_t, AAudioStream_getFormat, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_sharing_mode_t, AAudioStream_getSharingMode, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_performance_mode_t, AAudioStream_getPerformanceMode, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_direction_t, AAudioStream_getDirection, (AAudioStream * stream))
+SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesWritten, (AAudioStream * stream))
+SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesRead, (AAudioStream * stream))
+SDL_PROC_UNUSED(aaudio_session_id_t, AAudioStream_getSessionId, (AAudioStream * stream)) /* API 28 */
+SDL_PROC(aaudio_result_t, AAudioStream_getTimestamp, (AAudioStream * stream, clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds))
+SDL_PROC_UNUSED(aaudio_usage_t, AAudioStream_getUsage, (AAudioStream * stream))                                 /* API 28 */
+SDL_PROC_UNUSED(aaudio_content_type_t, AAudioStream_getContentType, (AAudioStream * stream))                    /* API 28 */
+SDL_PROC_UNUSED(aaudio_input_preset_t, AAudioStream_getInputPreset, (AAudioStream * stream))                    /* API 28 */
+SDL_PROC_UNUSED(aaudio_allowed_capture_policy_t, AAudioStream_getAllowedCapturePolicy, (AAudioStream * stream)) /* API 29 */
+SDL_PROC_UNUSED(bool, AAudioStream_isPrivacySensitive, (AAudioStream * stream))                                 /* API 30 */

third_party/SDL/src/audio/alsa/SDL_alsa_audio.c

@@ -0,0 +1,991 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_ALSA
+
+#ifndef SDL_ALSA_NON_BLOCKING
+#define SDL_ALSA_NON_BLOCKING 0
+#endif
+
+/* without the thread, you will detect devices on startup, but will not get futher hotplug events. But that might be okay. */
+#ifndef SDL_ALSA_HOTPLUG_THREAD
+#define SDL_ALSA_HOTPLUG_THREAD 1
+#endif
+
+/* Allow access to a raw mixing buffer */
+
+#include <sys/types.h>
+#include <signal.h> /* For kill() */
+#include <string.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_alsa_audio.h"
+
+#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
+#include "SDL_loadso.h"
+#endif
+
+static int (*ALSA_snd_pcm_open)(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
+static int (*ALSA_snd_pcm_close)(snd_pcm_t *pcm);
+static snd_pcm_sframes_t (*ALSA_snd_pcm_writei)(snd_pcm_t *, const void *, snd_pcm_uframes_t);
+static snd_pcm_sframes_t (*ALSA_snd_pcm_readi)(snd_pcm_t *, void *, snd_pcm_uframes_t);
+static int (*ALSA_snd_pcm_recover)(snd_pcm_t *, int, int);
+static int (*ALSA_snd_pcm_prepare)(snd_pcm_t *);
+static int (*ALSA_snd_pcm_drain)(snd_pcm_t *);
+static const char *(*ALSA_snd_strerror)(int);
+static size_t (*ALSA_snd_pcm_hw_params_sizeof)(void);
+static size_t (*ALSA_snd_pcm_sw_params_sizeof)(void);
+static void (*ALSA_snd_pcm_hw_params_copy)(snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);
+static int (*ALSA_snd_pcm_hw_params_any)(snd_pcm_t *, snd_pcm_hw_params_t *);
+static int (*ALSA_snd_pcm_hw_params_set_access)(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
+static int (*ALSA_snd_pcm_hw_params_set_format)(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
+static int (*ALSA_snd_pcm_hw_params_set_channels)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
+static int (*ALSA_snd_pcm_hw_params_get_channels)(const snd_pcm_hw_params_t *, unsigned int *);
+static int (*ALSA_snd_pcm_hw_params_set_rate_near)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
+static int (*ALSA_snd_pcm_hw_params_set_period_size_near)(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
+static int (*ALSA_snd_pcm_hw_params_get_period_size)(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
+static int (*ALSA_snd_pcm_hw_params_set_periods_min)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
+static int (*ALSA_snd_pcm_hw_params_set_periods_first)(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
+static int (*ALSA_snd_pcm_hw_params_get_periods)(const snd_pcm_hw_params_t *, unsigned int *, int *);
+static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)(snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
+static int (*ALSA_snd_pcm_hw_params_get_buffer_size)(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
+static int (*ALSA_snd_pcm_hw_params)(snd_pcm_t *, snd_pcm_hw_params_t *);
+static int (*ALSA_snd_pcm_sw_params_current)(snd_pcm_t *,
+                                             snd_pcm_sw_params_t *);
+static int (*ALSA_snd_pcm_sw_params_set_start_threshold)(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
+static int (*ALSA_snd_pcm_sw_params)(snd_pcm_t *, snd_pcm_sw_params_t *);
+static int (*ALSA_snd_pcm_nonblock)(snd_pcm_t *, int);
+static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
+static int (*ALSA_snd_pcm_sw_params_set_avail_min)(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
+static int (*ALSA_snd_pcm_reset)(snd_pcm_t *);
+static int (*ALSA_snd_device_name_hint)(int, const char *, void ***);
+static char *(*ALSA_snd_device_name_get_hint)(const void *, const char *);
+static int (*ALSA_snd_device_name_free_hint)(void **);
+static snd_pcm_sframes_t (*ALSA_snd_pcm_avail)(snd_pcm_t *);
+#ifdef SND_CHMAP_API_VERSION
+static snd_pcm_chmap_t *(*ALSA_snd_pcm_get_chmap)(snd_pcm_t *);
+static int (*ALSA_snd_pcm_chmap_print)(const snd_pcm_chmap_t *map, size_t maxlen, char *buf);
+#endif
+
+#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
+#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof
+#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof
+
+static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
+static void *alsa_handle = NULL;
+
+static int load_alsa_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(alsa_handle, fn);
+    if (*addr == NULL) {
+        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        return 0;
+    }
+
+    return 1;
+}
+
+/* cast funcs to char* first, to please GCC's strict aliasing rules. */
+#define SDL_ALSA_SYM(x)                                 \
+    if (!load_alsa_sym(#x, (void **)(char *)&ALSA_##x)) \
+    return -1
+#else
+#define SDL_ALSA_SYM(x) ALSA_##x = x
+#endif
+
+static int load_alsa_syms(void)
+{
+    SDL_ALSA_SYM(snd_pcm_open);
+    SDL_ALSA_SYM(snd_pcm_close);
+    SDL_ALSA_SYM(snd_pcm_writei);
+    SDL_ALSA_SYM(snd_pcm_readi);
+    SDL_ALSA_SYM(snd_pcm_recover);
+    SDL_ALSA_SYM(snd_pcm_prepare);
+    SDL_ALSA_SYM(snd_pcm_drain);
+    SDL_ALSA_SYM(snd_strerror);
+    SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);
+    SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);
+    SDL_ALSA_SYM(snd_pcm_hw_params_copy);
+    SDL_ALSA_SYM(snd_pcm_hw_params_any);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_access);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_format);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);
+    SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);
+    SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_min);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_first);
+    SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);
+    SDL_ALSA_SYM(snd_pcm_hw_params_set_buffer_size_near);
+    SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size);
+    SDL_ALSA_SYM(snd_pcm_hw_params);
+    SDL_ALSA_SYM(snd_pcm_sw_params_current);
+    SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);
+    SDL_ALSA_SYM(snd_pcm_sw_params);
+    SDL_ALSA_SYM(snd_pcm_nonblock);
+    SDL_ALSA_SYM(snd_pcm_wait);
+    SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
+    SDL_ALSA_SYM(snd_pcm_reset);
+    SDL_ALSA_SYM(snd_device_name_hint);
+    SDL_ALSA_SYM(snd_device_name_get_hint);
+    SDL_ALSA_SYM(snd_device_name_free_hint);
+    SDL_ALSA_SYM(snd_pcm_avail);
+#ifdef SND_CHMAP_API_VERSION
+    SDL_ALSA_SYM(snd_pcm_get_chmap);
+    SDL_ALSA_SYM(snd_pcm_chmap_print);
+#endif
+
+    return 0;
+}
+
+#undef SDL_ALSA_SYM
+
+#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
+
+static void UnloadALSALibrary(void)
+{
+    if (alsa_handle != NULL) {
+        SDL_UnloadObject(alsa_handle);
+        alsa_handle = NULL;
+    }
+}
+
+static int LoadALSALibrary(void)
+{
+    int retval = 0;
+    if (alsa_handle == NULL) {
+        alsa_handle = SDL_LoadObject(alsa_library);
+        if (alsa_handle == NULL) {
+            retval = -1;
+            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
+        } else {
+            retval = load_alsa_syms();
+            if (retval < 0) {
+                UnloadALSALibrary();
+            }
+        }
+    }
+    return retval;
+}
+
+#else
+
+static void UnloadALSALibrary(void)
+{
+}
+
+static int LoadALSALibrary(void)
+{
+    load_alsa_syms();
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
+
+static const char *get_audio_device(void *handle, const int channels)
+{
+    const char *device;
+
+    if (handle != NULL) {
+        return (const char *)handle;
+    }
+
+    /* !!! FIXME: we also check "SDL_AUDIO_DEVICE_NAME" at the higher level. */
+    device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
+    if (device != NULL) {
+        return device;
+    }
+
+    if (channels == 6) {
+        return "plug:surround51";
+    } else if (channels == 4) {
+        return "plug:surround40";
+    }
+
+    return "default";
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void ALSA_WaitDevice(_THIS)
+{
+#if SDL_ALSA_NON_BLOCKING
+    const snd_pcm_sframes_t needed = (snd_pcm_sframes_t)this->spec.samples;
+    while (SDL_AtomicGet(&this->enabled)) {
+        const snd_pcm_sframes_t rc = ALSA_snd_pcm_avail(this->hidden->pcm_handle);
+        if ((rc < 0) && (rc != -EAGAIN)) {
+            /* Hmm, not much we can do - abort */
+            fprintf(stderr, "ALSA snd_pcm_avail failed (unrecoverable): %s\n",
+                    ALSA_snd_strerror(rc));
+            SDL_OpenedAudioDeviceDisconnected(this);
+            return;
+        } else if (rc < needed) {
+            const Uint32 delay = ((needed - (SDL_max(rc, 0))) * 1000) / this->spec.freq;
+            SDL_Delay(SDL_max(delay, 10));
+        } else {
+            break; /* ready to go! */
+        }
+    }
+#endif
+}
+
+/* !!! FIXME: is there a channel swizzler in alsalib instead? */
+/*
+ * https://bugzilla.libsdl.org/show_bug.cgi?id=110
+ * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
+ *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
+ */
+#define SWIZ6(T)                                                                  \
+    static void swizzle_alsa_channels_6_##T(void *buffer, const Uint32 bufferlen) \
+    {                                                                             \
+        T *ptr = (T *)buffer;                                                     \
+        Uint32 i;                                                                 \
+        for (i = 0; i < bufferlen; i++, ptr += 6) {                               \
+            T tmp;                                                                \
+            tmp = ptr[2];                                                         \
+            ptr[2] = ptr[4];                                                      \
+            ptr[4] = tmp;                                                         \
+            tmp = ptr[3];                                                         \
+            ptr[3] = ptr[5];                                                      \
+            ptr[5] = tmp;                                                         \
+        }                                                                         \
+    }
+
+/* !!! FIXME: is there a channel swizzler in alsalib instead? */
+/* !!! FIXME: this screams for a SIMD shuffle operation. */
+/*
+ * https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/mapping-stream-formats-to-speaker-configurations
+ * For Linux ALSA, this appears to be FL-FR-RL-RR-C-LFE-SL-SR
+ *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-SL-SR-RL-RR"
+ */
+#define SWIZ8(T)                                                                  \
+    static void swizzle_alsa_channels_8_##T(void *buffer, const Uint32 bufferlen) \
+    {                                                                             \
+        T *ptr = (T *)buffer;                                                     \
+        Uint32 i;                                                                 \
+        for (i = 0; i < bufferlen; i++, ptr += 6) {                               \
+            const T center = ptr[2];                                              \
+            const T subwoofer = ptr[3];                                           \
+            const T side_left = ptr[4];                                           \
+            const T side_right = ptr[5];                                          \
+            const T rear_left = ptr[6];                                           \
+            const T rear_right = ptr[7];                                          \
+            ptr[2] = rear_left;                                                   \
+            ptr[3] = rear_right;                                                  \
+            ptr[4] = center;                                                      \
+            ptr[5] = subwoofer;                                                   \
+            ptr[6] = side_left;                                                   \
+            ptr[7] = side_right;                                                  \
+        }                                                                         \
+    }
+
+#define CHANNEL_SWIZZLE(x) \
+    x(Uint64)              \
+        x(Uint32)          \
+            x(Uint16)      \
+                x(Uint8)
+
+CHANNEL_SWIZZLE(SWIZ6)
+CHANNEL_SWIZZLE(SWIZ8)
+
+#undef CHANNEL_SWIZZLE
+#undef SWIZ6
+#undef SWIZ8
+
+/*
+ * Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
+ *  channels from Windows/Mac order to the format alsalib will want.
+ */
+static void swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
+{
+    switch (this->spec.channels) {
+#define CHANSWIZ(chans)                                                \
+    case chans:                                                        \
+        switch ((this->spec.format & (0xFF))) {                        \
+        case 8:                                                        \
+            swizzle_alsa_channels_##chans##_Uint8(buffer, bufferlen);  \
+            break;                                                     \
+        case 16:                                                       \
+            swizzle_alsa_channels_##chans##_Uint16(buffer, bufferlen); \
+            break;                                                     \
+        case 32:                                                       \
+            swizzle_alsa_channels_##chans##_Uint32(buffer, bufferlen); \
+            break;                                                     \
+        case 64:                                                       \
+            swizzle_alsa_channels_##chans##_Uint64(buffer, bufferlen); \
+            break;                                                     \
+        default:                                                       \
+            SDL_assert(!"unhandled bitsize");                          \
+            break;                                                     \
+        }                                                              \
+        return;
+
+        CHANSWIZ(6);
+        CHANSWIZ(8);
+#undef CHANSWIZ
+    default:
+        break;
+    }
+}
+
+#ifdef SND_CHMAP_API_VERSION
+/* Some devices have the right channel map, no swizzling necessary */
+static void no_swizzle(_THIS, void *buffer, Uint32 bufferlen)
+{
+}
+#endif /* SND_CHMAP_API_VERSION */
+
+static void ALSA_PlayDevice(_THIS)
+{
+    const Uint8 *sample_buf = (const Uint8 *)this->hidden->mixbuf;
+    const int frame_size = ((SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
+                           this->spec.channels;
+    snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t)this->spec.samples);
+
+    this->hidden->swizzle_func(this, this->hidden->mixbuf, frames_left);
+
+    while (frames_left > 0 && SDL_AtomicGet(&this->enabled)) {
+        int status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
+                                         sample_buf, frames_left);
+
+        if (status < 0) {
+            if (status == -EAGAIN) {
+                /* Apparently snd_pcm_recover() doesn't handle this case -
+                   does it assume snd_pcm_wait() above? */
+                SDL_Delay(1);
+                continue;
+            }
+            status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
+            if (status < 0) {
+                /* Hmm, not much we can do - abort */
+                SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
+                             "ALSA write failed (unrecoverable): %s\n",
+                             ALSA_snd_strerror(status));
+                SDL_OpenedAudioDeviceDisconnected(this);
+                return;
+            }
+            continue;
+        } else if (status == 0) {
+            /* No frames were written (no available space in pcm device).
+               Allow other threads to catch up. */
+            Uint32 delay = (frames_left / 2 * 1000) / this->spec.freq;
+            SDL_Delay(delay);
+        }
+
+        sample_buf += status * frame_size;
+        frames_left -= status;
+    }
+}
+
+static Uint8 *ALSA_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static int ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    Uint8 *sample_buf = (Uint8 *)buffer;
+    const int frame_size = ((SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
+                           this->spec.channels;
+    const int total_frames = buflen / frame_size;
+    snd_pcm_uframes_t frames_left = total_frames;
+    snd_pcm_uframes_t wait_time = frame_size / 2;
+
+    SDL_assert((buflen % frame_size) == 0);
+
+    while (frames_left > 0 && SDL_AtomicGet(&this->enabled)) {
+        int status;
+
+        status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
+                                    sample_buf, frames_left);
+
+        if (status == -EAGAIN) {
+            ALSA_snd_pcm_wait(this->hidden->pcm_handle, wait_time);
+            status = 0;
+        } else if (status < 0) {
+            /*printf("ALSA: capture error %d\n", status);*/
+            status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
+            if (status < 0) {
+                /* Hmm, not much we can do - abort */
+                SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
+                             "ALSA read failed (unrecoverable): %s\n",
+                             ALSA_snd_strerror(status));
+                return -1;
+            }
+            continue;
+        }
+
+        /*printf("ALSA: captured %d bytes\n", status * frame_size);*/
+        sample_buf += status * frame_size;
+        frames_left -= status;
+    }
+
+    this->hidden->swizzle_func(this, buffer, total_frames - frames_left);
+
+    return (total_frames - frames_left) * frame_size;
+}
+
+static void ALSA_FlushCapture(_THIS)
+{
+    ALSA_snd_pcm_reset(this->hidden->pcm_handle);
+}
+
+static void ALSA_CloseDevice(_THIS)
+{
+    if (this->hidden->pcm_handle) {
+        /* Wait for the submitted audio to drain
+           ALSA_snd_pcm_drop() can hang, so don't use that.
+         */
+        Uint32 delay = ((this->spec.samples * 1000) / this->spec.freq) * 2;
+        SDL_Delay(delay);
+
+        ALSA_snd_pcm_close(this->hidden->pcm_handle);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
+{
+    int status;
+    snd_pcm_hw_params_t *hwparams;
+    snd_pcm_uframes_t persize;
+    unsigned int periods;
+
+    /* Copy the hardware parameters for this setup */
+    snd_pcm_hw_params_alloca(&hwparams);
+    ALSA_snd_pcm_hw_params_copy(hwparams, params);
+
+    /* Attempt to match the period size to the requested buffer size */
+    persize = this->spec.samples;
+    status = ALSA_snd_pcm_hw_params_set_period_size_near(
+        this->hidden->pcm_handle, hwparams, &persize, NULL);
+    if (status < 0) {
+        return -1;
+    }
+
+    /* Need to at least double buffer */
+    periods = 2;
+    status = ALSA_snd_pcm_hw_params_set_periods_min(
+        this->hidden->pcm_handle, hwparams, &periods, NULL);
+    if (status < 0) {
+        return -1;
+    }
+
+    status = ALSA_snd_pcm_hw_params_set_periods_first(
+        this->hidden->pcm_handle, hwparams, &periods, NULL);
+    if (status < 0) {
+        return -1;
+    }
+
+    /* "set" the hardware with the desired parameters */
+    status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
+    if (status < 0) {
+        return -1;
+    }
+
+    this->spec.samples = persize;
+
+    /* This is useful for debugging */
+    if (SDL_getenv("SDL_AUDIO_ALSA_DEBUG")) {
+        snd_pcm_uframes_t bufsize;
+
+        ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
+
+        SDL_LogError(SDL_LOG_CATEGORY_AUDIO,
+                     "ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
+                     persize, periods, bufsize);
+    }
+
+    return 0;
+}
+
+static int ALSA_OpenDevice(_THIS, const char *devname)
+{
+    int status = 0;
+    SDL_bool iscapture = this->iscapture;
+    snd_pcm_t *pcm_handle = NULL;
+    snd_pcm_hw_params_t *hwparams = NULL;
+    snd_pcm_sw_params_t *swparams = NULL;
+    snd_pcm_format_t format = 0;
+    SDL_AudioFormat test_format = 0;
+    unsigned int rate = 0;
+    unsigned int channels = 0;
+#ifdef SND_CHMAP_API_VERSION
+    snd_pcm_chmap_t *chmap;
+    char chmap_str[64];
+#endif
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Open the audio device */
+    /* Name of device should depend on # channels in spec */
+    status = ALSA_snd_pcm_open(&pcm_handle,
+                               get_audio_device(this->handle, this->spec.channels),
+                               iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
+                               SND_PCM_NONBLOCK);
+
+    if (status < 0) {
+        return SDL_SetError("ALSA: Couldn't open audio device: %s", ALSA_snd_strerror(status));
+    }
+
+    this->hidden->pcm_handle = pcm_handle;
+
+    /* Figure out what the hardware is capable of */
+    snd_pcm_hw_params_alloca(&hwparams);
+    status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
+    if (status < 0) {
+        return SDL_SetError("ALSA: Couldn't get hardware config: %s", ALSA_snd_strerror(status));
+    }
+
+    /* SDL only uses interleaved sample output */
+    status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
+                                               SND_PCM_ACCESS_RW_INTERLEAVED);
+    if (status < 0) {
+        return SDL_SetError("ALSA: Couldn't set interleaved access: %s", ALSA_snd_strerror(status));
+    }
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_U8:
+            format = SND_PCM_FORMAT_U8;
+            break;
+        case AUDIO_S8:
+            format = SND_PCM_FORMAT_S8;
+            break;
+        case AUDIO_S16LSB:
+            format = SND_PCM_FORMAT_S16_LE;
+            break;
+        case AUDIO_S16MSB:
+            format = SND_PCM_FORMAT_S16_BE;
+            break;
+        case AUDIO_U16LSB:
+            format = SND_PCM_FORMAT_U16_LE;
+            break;
+        case AUDIO_U16MSB:
+            format = SND_PCM_FORMAT_U16_BE;
+            break;
+        case AUDIO_S32LSB:
+            format = SND_PCM_FORMAT_S32_LE;
+            break;
+        case AUDIO_S32MSB:
+            format = SND_PCM_FORMAT_S32_BE;
+            break;
+        case AUDIO_F32LSB:
+            format = SND_PCM_FORMAT_FLOAT_LE;
+            break;
+        case AUDIO_F32MSB:
+            format = SND_PCM_FORMAT_FLOAT_BE;
+            break;
+        default:
+            continue;
+        }
+        if (ALSA_snd_pcm_hw_params_set_format(pcm_handle, hwparams, format) >= 0) {
+            break;
+        }
+    }
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "alsa");
+    }
+    this->spec.format = test_format;
+
+    /* Validate number of channels and determine if swizzling is necessary
+     * Assume original swizzling, until proven otherwise.
+     */
+    this->hidden->swizzle_func = swizzle_alsa_channels;
+#ifdef SND_CHMAP_API_VERSION
+    chmap = ALSA_snd_pcm_get_chmap(pcm_handle);
+    if (chmap) {
+        if (ALSA_snd_pcm_chmap_print(chmap, sizeof(chmap_str), chmap_str) > 0) {
+            if (SDL_strcmp("FL FR FC LFE RL RR", chmap_str) == 0 ||
+                SDL_strcmp("FL FR FC LFE SL SR", chmap_str) == 0) {
+                this->hidden->swizzle_func = no_swizzle;
+            }
+        }
+        free(chmap);
+    }
+#endif /* SND_CHMAP_API_VERSION */
+
+    /* Set the number of channels */
+    status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
+                                                 this->spec.channels);
+    channels = this->spec.channels;
+    if (status < 0) {
+        status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
+        if (status < 0) {
+            return SDL_SetError("ALSA: Couldn't set audio channels");
+        }
+        this->spec.channels = channels;
+    }
+
+    /* Set the audio rate */
+    rate = this->spec.freq;
+    status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
+                                                  &rate, NULL);
+    if (status < 0) {
+        return SDL_SetError("ALSA: Couldn't set audio frequency: %s", ALSA_snd_strerror(status));
+    }
+    this->spec.freq = rate;
+
+    /* Set the buffer size, in samples */
+    status = ALSA_set_buffer_size(this, hwparams);
+    if (status < 0) {
+        return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
+    }
+
+    /* Set the software parameters */
+    snd_pcm_sw_params_alloca(&swparams);
+    status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
+    if (status < 0) {
+        return SDL_SetError("ALSA: Couldn't get software config: %s", ALSA_snd_strerror(status));
+    }
+    status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);
+    if (status < 0) {
+        return SDL_SetError("Couldn't set minimum available samples: %s", ALSA_snd_strerror(status));
+    }
+    status =
+        ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
+    if (status < 0) {
+        return SDL_SetError("ALSA: Couldn't set start threshold: %s", ALSA_snd_strerror(status));
+    }
+    status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
+    if (status < 0) {
+        return SDL_SetError("Couldn't set software audio parameters: %s", ALSA_snd_strerror(status));
+    }
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    if (!iscapture) {
+        this->hidden->mixlen = this->spec.size;
+        this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
+        if (this->hidden->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
+    }
+
+#if !SDL_ALSA_NON_BLOCKING
+    if (!iscapture) {
+        ALSA_snd_pcm_nonblock(pcm_handle, 0);
+    }
+#endif
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+typedef struct ALSA_Device
+{
+    char *name;
+    SDL_bool iscapture;
+    struct ALSA_Device *next;
+} ALSA_Device;
+
+static void add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSeen)
+{
+    ALSA_Device *dev = SDL_malloc(sizeof(ALSA_Device));
+    char *desc;
+    char *handle = NULL;
+    char *ptr;
+
+    if (dev == NULL) {
+        return;
+    }
+
+    /* Not all alsa devices are enumerable via snd_device_name_get_hint
+       (i.e. bluetooth devices).  Therefore if hint is passed in to this
+       function as  NULL, assume name contains desc.
+       Make sure not to free the storage associated with desc in this case */
+    if (hint) {
+        desc = ALSA_snd_device_name_get_hint(hint, "DESC");
+        if (desc == NULL) {
+            SDL_free(dev);
+            return;
+        }
+    } else {
+        desc = (char *)name;
+    }
+
+    SDL_assert(name != NULL);
+
+    /* some strings have newlines, like "HDA NVidia, HDMI 0\nHDMI Audio Output".
+       just chop the extra lines off, this seems to get a reasonable device
+       name without extra details. */
+    ptr = SDL_strchr(desc, '\n');
+    if (ptr != NULL) {
+        *ptr = '\0';
+    }
+
+    /*printf("ALSA: adding %s device '%s' (%s)\n", iscapture ? "capture" : "output", name, desc);*/
+
+    handle = SDL_strdup(name);
+    if (handle == NULL) {
+        if (hint) {
+            free(desc);
+        }
+        SDL_free(dev);
+        return;
+    }
+
+    /* Note that spec is NULL, because we are required to open the device before
+     * acquiring the mix format, making this information inaccessible at
+     * enumeration time
+     */
+    SDL_AddAudioDevice(iscapture, desc, NULL, handle);
+    if (hint) {
+        free(desc);
+    }
+    dev->name = handle;
+    dev->iscapture = iscapture;
+    dev->next = *pSeen;
+    *pSeen = dev;
+}
+
+static ALSA_Device *hotplug_devices = NULL;
+
+static void ALSA_HotplugIteration(void)
+{
+    void **hints = NULL;
+    ALSA_Device *dev;
+    ALSA_Device *unseen;
+    ALSA_Device *seen;
+    ALSA_Device *next;
+    ALSA_Device *prev;
+
+    if (ALSA_snd_device_name_hint(-1, "pcm", &hints) == 0) {
+        int i, j;
+        const char *match = NULL;
+        int bestmatch = 0xFFFF;
+        size_t match_len = 0;
+        int defaultdev = -1;
+        static const char *const prefixes[] = {
+            "hw:", "sysdefault:", "default:", NULL
+        };
+
+        unseen = hotplug_devices;
+        seen = NULL;
+
+        /* Apparently there are several different ways that ALSA lists
+           actual hardware. It could be prefixed with "hw:" or "default:"
+           or "sysdefault:" and maybe others. Go through the list and see
+           if we can find a preferred prefix for the system. */
+        for (i = 0; hints[i]; i++) {
+            char *name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
+            if (name == NULL) {
+                continue;
+            }
+
+            /* full name, not a prefix */
+            if ((defaultdev == -1) && (SDL_strcmp(name, "default") == 0)) {
+                defaultdev = i;
+            }
+
+            for (j = 0; prefixes[j]; j++) {
+                const char *prefix = prefixes[j];
+                const size_t prefixlen = SDL_strlen(prefix);
+                if (SDL_strncmp(name, prefix, prefixlen) == 0) {
+                    if (j < bestmatch) {
+                        bestmatch = j;
+                        match = prefix;
+                        match_len = prefixlen;
+                    }
+                }
+            }
+
+            free(name);
+        }
+
+        /* look through the list of device names to find matches */
+        for (i = 0; hints[i]; i++) {
+            char *name;
+
+            /* if we didn't find a device name prefix we like at all... */
+            if ((match == NULL) && (defaultdev != i)) {
+                continue; /* ...skip anything that isn't the default device. */
+            }
+
+            name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
+            if (name == NULL) {
+                continue;
+            }
+
+            /* only want physical hardware interfaces */
+            if (match == NULL || (SDL_strncmp(name, match, match_len) == 0)) {
+                char *ioid = ALSA_snd_device_name_get_hint(hints[i], "IOID");
+                const SDL_bool isoutput = (ioid == NULL) || (SDL_strcmp(ioid, "Output") == 0);
+                const SDL_bool isinput = (ioid == NULL) || (SDL_strcmp(ioid, "Input") == 0);
+                SDL_bool have_output = SDL_FALSE;
+                SDL_bool have_input = SDL_FALSE;
+
+                free(ioid);
+
+                if (!isoutput && !isinput) {
+                    free(name);
+                    continue;
+                }
+
+                prev = NULL;
+                for (dev = unseen; dev; dev = next) {
+                    next = dev->next;
+                    if ((SDL_strcmp(dev->name, name) == 0) && (((isinput) && dev->iscapture) || ((isoutput) && !dev->iscapture))) {
+                        if (prev) {
+                            prev->next = next;
+                        } else {
+                            unseen = next;
+                        }
+                        dev->next = seen;
+                        seen = dev;
+                        if (isinput) {
+                            have_input = SDL_TRUE;
+                        }
+                        if (isoutput) {
+                            have_output = SDL_TRUE;
+                        }
+                    } else {
+                        prev = dev;
+                    }
+                }
+
+                if (isinput && !have_input) {
+                    add_device(SDL_TRUE, name, hints[i], &seen);
+                }
+                if (isoutput && !have_output) {
+                    add_device(SDL_FALSE, name, hints[i], &seen);
+                }
+            }
+
+            free(name);
+        }
+
+        ALSA_snd_device_name_free_hint(hints);
+
+        hotplug_devices = seen; /* now we have a known-good list of attached devices. */
+
+        /* report anything still in unseen as removed. */
+        for (dev = unseen; dev; dev = next) {
+            /*printf("ALSA: removing usb %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/
+            next = dev->next;
+            SDL_RemoveAudioDevice(dev->iscapture, dev->name);
+            SDL_free(dev->name);
+            SDL_free(dev);
+        }
+    }
+}
+
+#if SDL_ALSA_HOTPLUG_THREAD
+static SDL_atomic_t ALSA_hotplug_shutdown;
+static SDL_Thread *ALSA_hotplug_thread;
+
+static int SDLCALL ALSA_HotplugThread(void *arg)
+{
+    SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
+
+    while (!SDL_AtomicGet(&ALSA_hotplug_shutdown)) {
+        /* Block awhile before checking again, unless we're told to stop. */
+        const Uint32 ticks = SDL_GetTicks() + 5000;
+        while (!SDL_AtomicGet(&ALSA_hotplug_shutdown) && !SDL_TICKS_PASSED(SDL_GetTicks(), ticks)) {
+            SDL_Delay(100);
+        }
+
+        ALSA_HotplugIteration(); /* run the check. */
+    }
+
+    return 0;
+}
+#endif
+
+static void ALSA_DetectDevices(void)
+{
+    ALSA_HotplugIteration(); /* run once now before a thread continues to check. */
+
+#if SDL_ALSA_HOTPLUG_THREAD
+    SDL_AtomicSet(&ALSA_hotplug_shutdown, 0);
+    ALSA_hotplug_thread = SDL_CreateThread(ALSA_HotplugThread, "SDLHotplugALSA", NULL);
+    /* if the thread doesn't spin, oh well, you just don't get further hotplug events. */
+#endif
+}
+
+static void ALSA_Deinitialize(void)
+{
+    ALSA_Device *dev;
+    ALSA_Device *next;
+
+#if SDL_ALSA_HOTPLUG_THREAD
+    if (ALSA_hotplug_thread != NULL) {
+        SDL_AtomicSet(&ALSA_hotplug_shutdown, 1);
+        SDL_WaitThread(ALSA_hotplug_thread, NULL);
+        ALSA_hotplug_thread = NULL;
+    }
+#endif
+
+    /* Shutting down! Clean up any data we've gathered. */
+    for (dev = hotplug_devices; dev; dev = next) {
+        /*printf("ALSA: at shutdown, removing %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/
+        next = dev->next;
+        SDL_free(dev->name);
+        SDL_free(dev);
+    }
+    hotplug_devices = NULL;
+
+    UnloadALSALibrary();
+}
+
+static SDL_bool ALSA_Init(SDL_AudioDriverImpl *impl)
+{
+    if (LoadALSALibrary() < 0) {
+        return SDL_FALSE;
+    }
+
+    /* Set the function pointers */
+    impl->DetectDevices = ALSA_DetectDevices;
+    impl->OpenDevice = ALSA_OpenDevice;
+    impl->WaitDevice = ALSA_WaitDevice;
+    impl->GetDeviceBuf = ALSA_GetDeviceBuf;
+    impl->PlayDevice = ALSA_PlayDevice;
+    impl->CloseDevice = ALSA_CloseDevice;
+    impl->Deinitialize = ALSA_Deinitialize;
+    impl->CaptureFromDevice = ALSA_CaptureFromDevice;
+    impl->FlushCapture = ALSA_FlushCapture;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->SupportsNonPow2Samples = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap ALSA_bootstrap = {
+    "alsa", "ALSA PCM audio", ALSA_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_ALSA */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/alsa/SDL_alsa_audio.h

@@ -0,0 +1,48 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_ALSA_audio_h_
+#define SDL_ALSA_audio_h_
+
+#include <alsa/asoundlib.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The audio device handle */
+    snd_pcm_t *pcm_handle;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    /* swizzle function */
+    void (*swizzle_func)(_THIS, void *buffer, Uint32 bufferlen);
+};
+
+#endif /* SDL_ALSA_audio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/android/SDL_androidaudio.c

@@ -0,0 +1,208 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_ANDROID
+
+/* Output audio to Android */
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_androidaudio.h"
+
+#include "../../core/android/SDL_android.h"
+
+#include <android/log.h>
+
+static SDL_AudioDevice *audioDevice = NULL;
+static SDL_AudioDevice *captureDevice = NULL;
+
+
+static int ANDROIDAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    SDL_AudioFormat test_format;
+    SDL_bool iscapture = this->iscapture;
+
+    SDL_assert((captureDevice == NULL) || !iscapture);
+    SDL_assert((audioDevice == NULL) || iscapture);
+
+    if (iscapture) {
+        captureDevice = this;
+    } else {
+        audioDevice = this;
+    }
+
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        if ((test_format == AUDIO_U8) ||
+            (test_format == AUDIO_S16) ||
+            (test_format == AUDIO_F32)) {
+            this->spec.format = test_format;
+            break;
+        }
+    }
+
+    if (!test_format) {
+        /* Didn't find a compatible format :( */
+        return SDL_SetError("%s: Unsupported audio format", "android");
+    }
+
+    {
+        int audio_device_id = 0;
+        if (devname != NULL) {
+            audio_device_id = SDL_atoi(devname);
+        }
+        if (Android_JNI_OpenAudioDevice(iscapture, audio_device_id, &this->spec) < 0) {
+            return -1;
+        }
+    }
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    return 0;
+}
+
+static void ANDROIDAUDIO_PlayDevice(_THIS)
+{
+    Android_JNI_WriteAudioBuffer();
+}
+
+static Uint8 *ANDROIDAUDIO_GetDeviceBuf(_THIS)
+{
+    return Android_JNI_GetAudioBuffer();
+}
+
+static int ANDROIDAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    return Android_JNI_CaptureAudioBuffer(buffer, buflen);
+}
+
+static void ANDROIDAUDIO_FlushCapture(_THIS)
+{
+    Android_JNI_FlushCapturedAudio();
+}
+
+static void ANDROIDAUDIO_CloseDevice(_THIS)
+{
+    /* At this point SDL_CloseAudioDevice via close_audio_device took care of terminating the audio thread
+       so it's safe to terminate the Java side buffer and AudioTrack
+     */
+    Android_JNI_CloseAudioDevice(this->iscapture);
+    if (this->iscapture) {
+        SDL_assert(captureDevice == this);
+        captureDevice = NULL;
+    } else {
+        SDL_assert(audioDevice == this);
+        audioDevice = NULL;
+    }
+    SDL_free(this->hidden);
+}
+
+static SDL_bool ANDROIDAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->DetectDevices = Android_DetectDevices;
+    impl->OpenDevice = ANDROIDAUDIO_OpenDevice;
+    impl->PlayDevice = ANDROIDAUDIO_PlayDevice;
+    impl->GetDeviceBuf = ANDROIDAUDIO_GetDeviceBuf;
+    impl->CloseDevice = ANDROIDAUDIO_CloseDevice;
+    impl->CaptureFromDevice = ANDROIDAUDIO_CaptureFromDevice;
+    impl->FlushCapture = ANDROIDAUDIO_FlushCapture;
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+
+    /* and the capabilities */
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->OnlyHasDefaultOutputDevice = SDL_FALSE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_FALSE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap ANDROIDAUDIO_bootstrap = {
+    "android", "SDL Android audio driver", ANDROIDAUDIO_Init, SDL_FALSE
+};
+
+/* Pause (block) all non already paused audio devices by taking their mixer lock */
+void ANDROIDAUDIO_PauseDevices(void)
+{
+    /* TODO: Handle multiple devices? */
+    struct SDL_PrivateAudioData *private;
+    if (audioDevice != NULL && audioDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
+        if (SDL_AtomicGet(&audioDevice->paused)) {
+            /* The device is already paused, leave it alone */
+            private->resume = SDL_FALSE;
+        } else {
+            SDL_LockMutex(audioDevice->mixer_lock);
+            SDL_AtomicSet(&audioDevice->paused, 1);
+            private->resume = SDL_TRUE;
+        }
+    }
+
+    if (captureDevice != NULL && captureDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
+        if (SDL_AtomicGet(&captureDevice->paused)) {
+            /* The device is already paused, leave it alone */
+            private->resume = SDL_FALSE;
+        } else {
+            SDL_LockMutex(captureDevice->mixer_lock);
+            SDL_AtomicSet(&captureDevice->paused, 1);
+            private->resume = SDL_TRUE;
+        }
+    }
+}
+
+/* Resume (unblock) all non already paused audio devices by releasing their mixer lock */
+void ANDROIDAUDIO_ResumeDevices(void)
+{
+    /* TODO: Handle multiple devices? */
+    struct SDL_PrivateAudioData *private;
+    if (audioDevice != NULL && audioDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)audioDevice->hidden;
+        if (private->resume) {
+            SDL_AtomicSet(&audioDevice->paused, 0);
+            private->resume = SDL_FALSE;
+            SDL_UnlockMutex(audioDevice->mixer_lock);
+        }
+    }
+
+    if (captureDevice != NULL && captureDevice->hidden != NULL) {
+        private = (struct SDL_PrivateAudioData *)captureDevice->hidden;
+        if (private->resume) {
+            SDL_AtomicSet(&captureDevice->paused, 0);
+            private->resume = SDL_FALSE;
+            SDL_UnlockMutex(captureDevice->mixer_lock);
+        }
+    }
+}
+
+#else
+
+void ANDROIDAUDIO_ResumeDevices(void) {}
+void ANDROIDAUDIO_PauseDevices(void) {}
+
+#endif /* SDL_AUDIO_DRIVER_ANDROID */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/android/SDL_androidaudio.h

@@ -0,0 +1,42 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_androidaudio_h_
+#define SDL_androidaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* Resume device if it was paused automatically */
+    int resume;
+};
+
+void ANDROIDAUDIO_ResumeDevices(void);
+void ANDROIDAUDIO_PauseDevices(void);
+
+#endif /* SDL_androidaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/arts/SDL_artsaudio.c

@@ -0,0 +1,344 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_ARTS
+
+/* Allow access to a raw mixing buffer */
+
+#ifdef HAVE_SIGNAL_H
+#include <signal.h>
+#endif
+#include <unistd.h>
+#include <errno.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_artsaudio.h"
+
+#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
+#include "SDL_name.h"
+#include "SDL_loadso.h"
+#else
+#define SDL_NAME(X) X
+#endif
+
+#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
+
+static const char *arts_library = SDL_AUDIO_DRIVER_ARTS_DYNAMIC;
+static void *arts_handle = NULL;
+
+/* !!! FIXME: I hate this SDL_NAME clutter...it makes everything so messy! */
+static int (*SDL_NAME(arts_init)) (void);
+static void (*SDL_NAME(arts_free)) (void);
+static arts_stream_t(*SDL_NAME(arts_play_stream)) (int rate, int bits,
+                                                   int channels,
+                                                   const char *name);
+static int (*SDL_NAME(arts_stream_set)) (arts_stream_t s,
+                                         arts_parameter_t param, int value);
+static int (*SDL_NAME(arts_stream_get)) (arts_stream_t s,
+                                         arts_parameter_t param);
+static int (*SDL_NAME(arts_write)) (arts_stream_t s, const void *buffer,
+                                    int count);
+static void (*SDL_NAME(arts_close_stream)) (arts_stream_t s);
+static int (*SDL_NAME(arts_suspend))(void);
+static int (*SDL_NAME(arts_suspended)) (void);
+static const char *(*SDL_NAME(arts_error_text)) (int errorcode);
+
+#define SDL_ARTS_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
+static struct
+{
+    const char *name;
+    void **func;
+} arts_functions[] = {
+/* *INDENT-OFF* */
+    SDL_ARTS_SYM(arts_init),
+    SDL_ARTS_SYM(arts_free),
+    SDL_ARTS_SYM(arts_play_stream),
+    SDL_ARTS_SYM(arts_stream_set),
+    SDL_ARTS_SYM(arts_stream_get),
+    SDL_ARTS_SYM(arts_write),
+    SDL_ARTS_SYM(arts_close_stream),
+    SDL_ARTS_SYM(arts_suspend),
+    SDL_ARTS_SYM(arts_suspended),
+    SDL_ARTS_SYM(arts_error_text),
+/* *INDENT-ON* */
+};
+
+#undef SDL_ARTS_SYM
+
+static void UnloadARTSLibrary()
+{
+    if (arts_handle != NULL) {
+        SDL_UnloadObject(arts_handle);
+        arts_handle = NULL;
+    }
+}
+
+static int LoadARTSLibrary(void)
+{
+    int i, retval = -1;
+
+    if (arts_handle == NULL) {
+        arts_handle = SDL_LoadObject(arts_library);
+        if (arts_handle != NULL) {
+            retval = 0;
+            for (i = 0; i < SDL_arraysize(arts_functions); ++i) {
+                *arts_functions[i].func =
+                    SDL_LoadFunction(arts_handle, arts_functions[i].name);
+                if (!*arts_functions[i].func) {
+                    retval = -1;
+                    UnloadARTSLibrary();
+                    break;
+                }
+            }
+        }
+    }
+
+    return retval;
+}
+
+#else
+
+static void UnloadARTSLibrary()
+{
+    return;
+}
+
+static int LoadARTSLibrary(void)
+{
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_ARTS_DYNAMIC */
+
+/* This function waits until it is possible to write a full sound buffer */
+static void ARTS_WaitDevice(_THIS)
+{
+    Sint32 ticks;
+
+    /* Check to see if the thread-parent process is still alive */
+    {
+        static int cnt = 0;
+        /* Note that this only works with thread implementations
+           that use a different process id for each thread.
+         */
+        /* Check every 10 loops */
+        if (this->hidden->parent && (((++cnt) % 10) == 0)) {
+            if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
+                SDL_OpenedAudioDeviceDisconnected(this);
+            }
+        }
+    }
+
+    /* Use timer for general audio synchronization */
+    ticks =
+        ((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
+    if (ticks > 0) {
+        SDL_Delay(ticks);
+    }
+}
+
+static void ARTS_PlayDevice(_THIS)
+{
+    /* Write the audio data */
+    int written = SDL_NAME(arts_write) (this->hidden->stream,
+                                        this->hidden->mixbuf,
+                                        this->hidden->mixlen);
+
+    /* If timer synchronization is enabled, set the next write frame */
+    if (this->hidden->frame_ticks) {
+        this->hidden->next_frame += this->hidden->frame_ticks;
+    }
+
+    /* If we couldn't write, assume fatal error for now */
+    if (written < 0) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+}
+
+static Uint8 *ARTS_GetDeviceBuf(_THIS)
+{
+    return (this->hidden->mixbuf);
+}
+
+
+static void ARTS_CloseDevice(_THIS)
+{
+    if (this->hidden->stream) {
+        SDL_NAME(arts_close_stream) (this->hidden->stream);
+    }
+    SDL_NAME(arts_free) ();
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int ARTS_Suspend(void)
+{
+    const Uint32 abortms = SDL_GetTicks() + 3000; /* give up after 3 secs */
+    while ( (!SDL_NAME(arts_suspended)()) && !SDL_TICKS_PASSED(SDL_GetTicks(), abortms) ) {
+        if ( SDL_NAME(arts_suspend)() ) {
+            break;
+        }
+    }
+    return SDL_NAME(arts_suspended)();
+}
+
+static int ARTS_OpenDevice(_THIS, const char *devname)
+{
+    int rc = 0;
+    int bits, frag_spec = 0;
+    SDL_AudioFormat test_format = 0;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        switch (test_format) {
+        case AUDIO_U8:
+        case AUDIO_S16LSB:
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "arts");
+    }
+    this->spec.format = test_format;
+    bits = SDL_AUDIO_BITSIZE(test_format);
+
+    if ((rc = SDL_NAME(arts_init) ()) != 0) {
+        return SDL_SetError("Unable to initialize ARTS: %s",
+                            SDL_NAME(arts_error_text) (rc));
+    }
+
+    if (!ARTS_Suspend()) {
+        return SDL_SetError("ARTS can not open audio device");
+    }
+
+    this->hidden->stream = SDL_NAME(arts_play_stream) (this->spec.freq,
+                                                       bits,
+                                                       this->spec.channels,
+                                                       "SDL");
+
+    /* Play nothing so we have at least one write (server bug workaround). */
+    SDL_NAME(arts_write) (this->hidden->stream, "", 0);
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Determine the power of two of the fragment size */
+    for (frag_spec = 0; (0x01 << frag_spec) < this->spec.size; ++frag_spec);
+    if ((0x01 << frag_spec) != this->spec.size) {
+        return SDL_SetError("Fragment size must be a power of two");
+    }
+    frag_spec |= 0x00020000;    /* two fragments, for low latency */
+
+#ifdef ARTS_P_PACKET_SETTINGS
+    SDL_NAME(arts_stream_set) (this->hidden->stream,
+                               ARTS_P_PACKET_SETTINGS, frag_spec);
+#else
+    SDL_NAME(arts_stream_set) (this->hidden->stream, ARTS_P_PACKET_SIZE,
+                               frag_spec & 0xffff);
+    SDL_NAME(arts_stream_set) (this->hidden->stream, ARTS_P_PACKET_COUNT,
+                               frag_spec >> 16);
+#endif
+    this->spec.size = SDL_NAME(arts_stream_get) (this->hidden->stream,
+                                                 ARTS_P_PACKET_SIZE);
+
+    /* Allocate mixing buffer */
+    this->hidden->mixlen = this->spec.size;
+    this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+
+    /* Get the parent process id (we're the parent of the audio thread) */
+    this->hidden->parent = getpid();
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+
+static void ARTS_Deinitialize(void)
+{
+    UnloadARTSLibrary();
+}
+
+
+static SDL_bool ARTS_Init(SDL_AudioDriverImpl * impl)
+{
+    if (LoadARTSLibrary() < 0) {
+        return SDL_FALSE;
+    } else {
+        if (SDL_NAME(arts_init) () != NULL) {
+            UnloadARTSLibrary();
+            SDL_SetError("ARTS: arts_init failed (no audio server?)");
+            return SDL_FALSE;
+        }
+
+        /* Play a stream so aRts doesn't crash */
+        if (ARTS_Suspend()) {
+            arts_stream_t stream;
+            stream = SDL_NAME(arts_play_stream) (44100, 16, 2, "SDL");
+            SDL_NAME(arts_write) (stream, "", 0);
+            SDL_NAME(arts_close_stream) (stream);
+        }
+
+        SDL_NAME(arts_free) ();
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = ARTS_OpenDevice;
+    impl->PlayDevice = ARTS_PlayDevice;
+    impl->WaitDevice = ARTS_WaitDevice;
+    impl->GetDeviceBuf = ARTS_GetDeviceBuf;
+    impl->CloseDevice = ARTS_CloseDevice;
+    impl->Deinitialize = ARTS_Deinitialize;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+
+AudioBootStrap ARTS_bootstrap = {
+    "arts", "Analog RealTime Synthesizer", ARTS_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_ARTS */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/arts/SDL_artsaudio.h

@@ -0,0 +1,53 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_artsaudio_h_
+#define SDL_artsaudio_h_
+
+#include <artsc.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The stream descriptor for the audio device */
+    arts_stream_t stream;
+
+    /* The parent process id, to detect when application quits */
+    pid_t parent;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    /* Support for audio timing using a timer, in addition to SDL_IOReady() */
+    float frame_ticks;
+    float next_frame;
+};
+#define FUDGE_TICKS 10      /* The scheduler overhead ticks per frame */
+
+#endif /* SDL_artsaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/coreaudio/SDL_coreaudio.h

@@ -0,0 +1,76 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_coreaudio_h_
+#define SDL_coreaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+#if !defined(__IPHONEOS__)
+#define MACOSX_COREAUDIO 1
+#endif
+
+#if MACOSX_COREAUDIO
+#include <CoreAudio/CoreAudio.h>
+#else
+#import <AVFoundation/AVFoundation.h>
+#import <UIKit/UIApplication.h>
+#endif
+
+#include <AudioToolbox/AudioToolbox.h>
+#include <AudioUnit/AudioUnit.h>
+
+/* Things named "Master" were renamed to "Main" in macOS 12.0's SDK. */
+#if MACOSX_COREAUDIO
+#include <AvailabilityMacros.h>
+#ifndef MAC_OS_VERSION_12_0
+#define kAudioObjectPropertyElementMain kAudioObjectPropertyElementMaster
+#endif
+#endif
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    SDL_Thread *thread;
+    AudioQueueRef audioQueue;
+    int numAudioBuffers;
+    AudioQueueBufferRef *audioBuffer;
+    void *buffer;
+    UInt32 bufferOffset;
+    UInt32 bufferSize;
+    AudioStreamBasicDescription strdesc;
+    SDL_sem *ready_semaphore;
+    char *thread_error;
+#if MACOSX_COREAUDIO
+    AudioDeviceID deviceID;
+    SDL_atomic_t device_change_flag;
+#else
+    SDL_bool interrupted;
+    CFTypeRef interruption_listener;
+#endif
+};
+
+#endif /* SDL_coreaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/directsound/SDL_directsound.c

@@ -0,0 +1,659 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_DSOUND
+
+/* Allow access to a raw mixing buffer */
+
+#include "SDL_timer.h"
+#include "SDL_loadso.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_directsound.h"
+#include <mmreg.h>
+#if HAVE_MMDEVICEAPI_H
+#include "../../core/windows/SDL_immdevice.h"
+#endif /* HAVE_MMDEVICEAPI_H */
+
+#ifndef WAVE_FORMAT_IEEE_FLOAT
+#define WAVE_FORMAT_IEEE_FLOAT 0x0003
+#endif
+
+/* For Vista+, we can enumerate DSound devices with IMMDevice */
+#if HAVE_MMDEVICEAPI_H
+static SDL_bool SupportsIMMDevice = SDL_FALSE;
+#endif /* HAVE_MMDEVICEAPI_H */
+
+/* DirectX function pointers for audio */
+static void *DSoundDLL = NULL;
+typedef HRESULT(WINAPI *fnDirectSoundCreate8)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
+typedef HRESULT(WINAPI *fnDirectSoundEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
+typedef HRESULT(WINAPI *fnDirectSoundCaptureCreate8)(LPCGUID, LPDIRECTSOUNDCAPTURE8 *, LPUNKNOWN);
+typedef HRESULT(WINAPI *fnDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW, LPVOID);
+static fnDirectSoundCreate8 pDirectSoundCreate8 = NULL;
+static fnDirectSoundEnumerateW pDirectSoundEnumerateW = NULL;
+static fnDirectSoundCaptureCreate8 pDirectSoundCaptureCreate8 = NULL;
+static fnDirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW = NULL;
+
+static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
+static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
+
+static void DSOUND_Unload(void)
+{
+    pDirectSoundCreate8 = NULL;
+    pDirectSoundEnumerateW = NULL;
+    pDirectSoundCaptureCreate8 = NULL;
+    pDirectSoundCaptureEnumerateW = NULL;
+
+    if (DSoundDLL != NULL) {
+        SDL_UnloadObject(DSoundDLL);
+        DSoundDLL = NULL;
+    }
+}
+
+static int DSOUND_Load(void)
+{
+    int loaded = 0;
+
+    DSOUND_Unload();
+
+    DSoundDLL = SDL_LoadObject("DSOUND.DLL");
+    if (DSoundDLL == NULL) {
+        SDL_SetError("DirectSound: failed to load DSOUND.DLL");
+    } else {
+/* Now make sure we have DirectX 8 or better... */
+#define DSOUNDLOAD(f)                                  \
+    {                                                  \
+        p##f = (fn##f)SDL_LoadFunction(DSoundDLL, #f); \
+        if (!p##f)                                     \
+            loaded = 0;                                \
+    }
+        loaded = 1; /* will reset if necessary. */
+        DSOUNDLOAD(DirectSoundCreate8);
+        DSOUNDLOAD(DirectSoundEnumerateW);
+        DSOUNDLOAD(DirectSoundCaptureCreate8);
+        DSOUNDLOAD(DirectSoundCaptureEnumerateW);
+#undef DSOUNDLOAD
+
+        if (!loaded) {
+            SDL_SetError("DirectSound: System doesn't appear to have DX8.");
+        }
+    }
+
+    if (!loaded) {
+        DSOUND_Unload();
+    }
+
+    return loaded;
+}
+
+static int SetDSerror(const char *function, int code)
+{
+    const char *error;
+
+    switch (code) {
+    case E_NOINTERFACE:
+        error = "Unsupported interface -- Is DirectX 8.0 or later installed?";
+        break;
+    case DSERR_ALLOCATED:
+        error = "Audio device in use";
+        break;
+    case DSERR_BADFORMAT:
+        error = "Unsupported audio format";
+        break;
+    case DSERR_BUFFERLOST:
+        error = "Mixing buffer was lost";
+        break;
+    case DSERR_CONTROLUNAVAIL:
+        error = "Control requested is not available";
+        break;
+    case DSERR_INVALIDCALL:
+        error = "Invalid call for the current state";
+        break;
+    case DSERR_INVALIDPARAM:
+        error = "Invalid parameter";
+        break;
+    case DSERR_NODRIVER:
+        error = "No audio device found";
+        break;
+    case DSERR_OUTOFMEMORY:
+        error = "Out of memory";
+        break;
+    case DSERR_PRIOLEVELNEEDED:
+        error = "Caller doesn't have priority";
+        break;
+    case DSERR_UNSUPPORTED:
+        error = "Function not supported";
+        break;
+    default:
+        error = "Unknown DirectSound error";
+        break;
+    }
+
+    return SDL_SetError("%s: %s (0x%x)", function, error, code);
+}
+
+static void DSOUND_FreeDeviceHandle(void *handle)
+{
+    SDL_free(handle);
+}
+
+static int DSOUND_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
+{
+#if HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        return SDL_IMMDevice_GetDefaultAudioInfo(name, spec, iscapture);
+    }
+#endif /* HAVE_MMDEVICEAPI_H */
+    return SDL_Unsupported();
+}
+
+static BOOL CALLBACK FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID data)
+{
+    const int iscapture = (int)((size_t)data);
+    if (guid != NULL) { /* skip default device */
+        char *str = WIN_LookupAudioDeviceName(desc, guid);
+        if (str != NULL) {
+            LPGUID cpyguid = (LPGUID)SDL_malloc(sizeof(GUID));
+            SDL_memcpy(cpyguid, guid, sizeof(GUID));
+
+            /* Note that spec is NULL, because we are required to connect to the
+             * device before getting the channel mask and output format, making
+             * this information inaccessible at enumeration time
+             */
+            SDL_AddAudioDevice(iscapture, str, NULL, cpyguid);
+            SDL_free(str); /* addfn() makes a copy of this string. */
+        }
+    }
+    return TRUE; /* keep enumerating. */
+}
+
+static void DSOUND_DetectDevices(void)
+{
+#if HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        SDL_IMMDevice_EnumerateEndpoints(SDL_TRUE);
+    } else {
+#endif /* HAVE_MMDEVICEAPI_H */
+        pDirectSoundCaptureEnumerateW(FindAllDevs, (void *)((size_t)1));
+        pDirectSoundEnumerateW(FindAllDevs, (void *)((size_t)0));
+#if HAVE_MMDEVICEAPI_H
+    }
+#endif /* HAVE_MMDEVICEAPI_H*/
+}
+
+static void DSOUND_WaitDevice(_THIS)
+{
+    DWORD status = 0;
+    DWORD cursor = 0;
+    DWORD junk = 0;
+    HRESULT result = DS_OK;
+
+    /* Semi-busy wait, since we have no way of getting play notification
+       on a primary mixing buffer located in hardware (DirectX 5.0)
+     */
+    result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
+                                                   &junk, &cursor);
+    if (result != DS_OK) {
+        if (result == DSERR_BUFFERLOST) {
+            IDirectSoundBuffer_Restore(this->hidden->mixbuf);
+        }
+#ifdef DEBUG_SOUND
+        SetDSerror("DirectSound GetCurrentPosition", result);
+#endif
+        return;
+    }
+
+    while ((cursor / this->spec.size) == this->hidden->lastchunk) {
+        /* FIXME: find out how much time is left and sleep that long */
+        SDL_Delay(1);
+
+        /* Try to restore a lost sound buffer */
+        IDirectSoundBuffer_GetStatus(this->hidden->mixbuf, &status);
+        if (status & DSBSTATUS_BUFFERLOST) {
+            IDirectSoundBuffer_Restore(this->hidden->mixbuf);
+            IDirectSoundBuffer_GetStatus(this->hidden->mixbuf, &status);
+            if (status & DSBSTATUS_BUFFERLOST) {
+                break;
+            }
+        }
+        if (!(status & DSBSTATUS_PLAYING)) {
+            result = IDirectSoundBuffer_Play(this->hidden->mixbuf, 0, 0,
+                                             DSBPLAY_LOOPING);
+            if (result == DS_OK) {
+                continue;
+            }
+#ifdef DEBUG_SOUND
+            SetDSerror("DirectSound Play", result);
+#endif
+            return;
+        }
+
+        /* Find out where we are playing */
+        result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
+                                                       &junk, &cursor);
+        if (result != DS_OK) {
+            SetDSerror("DirectSound GetCurrentPosition", result);
+            return;
+        }
+    }
+}
+
+static void DSOUND_PlayDevice(_THIS)
+{
+    /* Unlock the buffer, allowing it to play */
+    if (this->hidden->locked_buf) {
+        IDirectSoundBuffer_Unlock(this->hidden->mixbuf,
+                                  this->hidden->locked_buf,
+                                  this->spec.size, NULL, 0);
+    }
+}
+
+static Uint8 *DSOUND_GetDeviceBuf(_THIS)
+{
+    DWORD cursor = 0;
+    DWORD junk = 0;
+    HRESULT result = DS_OK;
+    DWORD rawlen = 0;
+
+    /* Figure out which blocks to fill next */
+    this->hidden->locked_buf = NULL;
+    result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
+                                                   &junk, &cursor);
+    if (result == DSERR_BUFFERLOST) {
+        IDirectSoundBuffer_Restore(this->hidden->mixbuf);
+        result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
+                                                       &junk, &cursor);
+    }
+    if (result != DS_OK) {
+        SetDSerror("DirectSound GetCurrentPosition", result);
+        return NULL;
+    }
+    cursor /= this->spec.size;
+#ifdef DEBUG_SOUND
+    /* Detect audio dropouts */
+    {
+        DWORD spot = cursor;
+        if (spot < this->hidden->lastchunk) {
+            spot += this->hidden->num_buffers;
+        }
+        if (spot > this->hidden->lastchunk + 1) {
+            fprintf(stderr, "Audio dropout, missed %d fragments\n",
+                    (spot - (this->hidden->lastchunk + 1)));
+        }
+    }
+#endif
+    this->hidden->lastchunk = cursor;
+    cursor = (cursor + 1) % this->hidden->num_buffers;
+    cursor *= this->spec.size;
+
+    /* Lock the audio buffer */
+    result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
+                                     this->spec.size,
+                                     (LPVOID *)&this->hidden->locked_buf,
+                                     &rawlen, NULL, &junk, 0);
+    if (result == DSERR_BUFFERLOST) {
+        IDirectSoundBuffer_Restore(this->hidden->mixbuf);
+        result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
+                                         this->spec.size,
+                                         (LPVOID *)&this->hidden->locked_buf, &rawlen, NULL,
+                                         &junk, 0);
+    }
+    if (result != DS_OK) {
+        SetDSerror("DirectSound Lock", result);
+        return NULL;
+    }
+    return this->hidden->locked_buf;
+}
+
+static int DSOUND_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    DWORD junk, cursor, ptr1len, ptr2len;
+    VOID *ptr1, *ptr2;
+
+    SDL_assert(buflen == this->spec.size);
+
+    while (SDL_TRUE) {
+        if (SDL_AtomicGet(&this->shutdown)) { /* in case the buffer froze... */
+            SDL_memset(buffer, this->spec.silence, buflen);
+            return buflen;
+        }
+
+        if (IDirectSoundCaptureBuffer_GetCurrentPosition(h->capturebuf, &junk, &cursor) != DS_OK) {
+            return -1;
+        }
+        if ((cursor / this->spec.size) == h->lastchunk) {
+            SDL_Delay(1); /* FIXME: find out how much time is left and sleep that long */
+        } else {
+            break;
+        }
+    }
+
+    if (IDirectSoundCaptureBuffer_Lock(h->capturebuf, h->lastchunk * this->spec.size, this->spec.size, &ptr1, &ptr1len, &ptr2, &ptr2len, 0) != DS_OK) {
+        return -1;
+    }
+
+    SDL_assert(ptr1len == this->spec.size);
+    SDL_assert(ptr2 == NULL);
+    SDL_assert(ptr2len == 0);
+
+    SDL_memcpy(buffer, ptr1, ptr1len);
+
+    if (IDirectSoundCaptureBuffer_Unlock(h->capturebuf, ptr1, ptr1len, ptr2, ptr2len) != DS_OK) {
+        return -1;
+    }
+
+    h->lastchunk = (h->lastchunk + 1) % h->num_buffers;
+
+    return ptr1len;
+}
+
+static void DSOUND_FlushCapture(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    DWORD junk, cursor;
+    if (IDirectSoundCaptureBuffer_GetCurrentPosition(h->capturebuf, &junk, &cursor) == DS_OK) {
+        h->lastchunk = cursor / this->spec.size;
+    }
+}
+
+static void DSOUND_CloseDevice(_THIS)
+{
+    if (this->hidden->mixbuf != NULL) {
+        IDirectSoundBuffer_Stop(this->hidden->mixbuf);
+        IDirectSoundBuffer_Release(this->hidden->mixbuf);
+    }
+    if (this->hidden->sound != NULL) {
+        IDirectSound_Release(this->hidden->sound);
+    }
+    if (this->hidden->capturebuf != NULL) {
+        IDirectSoundCaptureBuffer_Stop(this->hidden->capturebuf);
+        IDirectSoundCaptureBuffer_Release(this->hidden->capturebuf);
+    }
+    if (this->hidden->capture != NULL) {
+        IDirectSoundCapture_Release(this->hidden->capture);
+    }
+    SDL_free(this->hidden);
+}
+
+/* This function tries to create a secondary audio buffer, and returns the
+   number of audio chunks available in the created buffer. This is for
+   playback devices, not capture.
+*/
+static int CreateSecondary(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
+{
+    LPDIRECTSOUND sndObj = this->hidden->sound;
+    LPDIRECTSOUNDBUFFER *sndbuf = &this->hidden->mixbuf;
+    HRESULT result = DS_OK;
+    DSBUFFERDESC format;
+    LPVOID pvAudioPtr1, pvAudioPtr2;
+    DWORD dwAudioBytes1, dwAudioBytes2;
+
+    /* Try to create the secondary buffer */
+    SDL_zero(format);
+    format.dwSize = sizeof(format);
+    format.dwFlags = DSBCAPS_GETCURRENTPOSITION2;
+    format.dwFlags |= DSBCAPS_GLOBALFOCUS;
+    format.dwBufferBytes = bufsize;
+    format.lpwfxFormat = wfmt;
+    result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
+    if (result != DS_OK) {
+        return SetDSerror("DirectSound CreateSoundBuffer", result);
+    }
+    IDirectSoundBuffer_SetFormat(*sndbuf, wfmt);
+
+    /* Silence the initial audio buffer */
+    result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
+                                     (LPVOID *)&pvAudioPtr1, &dwAudioBytes1,
+                                     (LPVOID *)&pvAudioPtr2, &dwAudioBytes2,
+                                     DSBLOCK_ENTIREBUFFER);
+    if (result == DS_OK) {
+        SDL_memset(pvAudioPtr1, this->spec.silence, dwAudioBytes1);
+        IDirectSoundBuffer_Unlock(*sndbuf,
+                                  (LPVOID)pvAudioPtr1, dwAudioBytes1,
+                                  (LPVOID)pvAudioPtr2, dwAudioBytes2);
+    }
+
+    /* We're ready to go */
+    return 0;
+}
+
+/* This function tries to create a capture buffer, and returns the
+   number of audio chunks available in the created buffer. This is for
+   capture devices, not playback.
+*/
+static int CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
+{
+    LPDIRECTSOUNDCAPTURE capture = this->hidden->capture;
+    LPDIRECTSOUNDCAPTUREBUFFER *capturebuf = &this->hidden->capturebuf;
+    DSCBUFFERDESC format;
+    HRESULT result;
+
+    SDL_zero(format);
+    format.dwSize = sizeof(format);
+    format.dwFlags = DSCBCAPS_WAVEMAPPED;
+    format.dwBufferBytes = bufsize;
+    format.lpwfxFormat = wfmt;
+
+    result = IDirectSoundCapture_CreateCaptureBuffer(capture, &format, capturebuf, NULL);
+    if (result != DS_OK) {
+        return SetDSerror("DirectSound CreateCaptureBuffer", result);
+    }
+
+    result = IDirectSoundCaptureBuffer_Start(*capturebuf, DSCBSTART_LOOPING);
+    if (result != DS_OK) {
+        IDirectSoundCaptureBuffer_Release(*capturebuf);
+        return SetDSerror("DirectSound Start", result);
+    }
+
+#if 0
+    /* presumably this starts at zero, but just in case... */
+    result = IDirectSoundCaptureBuffer_GetCurrentPosition(*capturebuf, &junk, &cursor);
+    if (result != DS_OK) {
+        IDirectSoundCaptureBuffer_Stop(*capturebuf);
+        IDirectSoundCaptureBuffer_Release(*capturebuf);
+        return SetDSerror("DirectSound GetCurrentPosition", result);
+    }
+
+    this->hidden->lastchunk = cursor / this->spec.size;
+#endif
+
+    return 0;
+}
+
+static int DSOUND_OpenDevice(_THIS, const char *devname)
+{
+    const DWORD numchunks = 8;
+    HRESULT result;
+    SDL_bool tried_format = SDL_FALSE;
+    SDL_bool iscapture = this->iscapture;
+    SDL_AudioFormat test_format;
+    LPGUID guid = (LPGUID)this->handle;
+    DWORD bufsize;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Open the audio device */
+    if (iscapture) {
+        result = pDirectSoundCaptureCreate8(guid, &this->hidden->capture, NULL);
+        if (result != DS_OK) {
+            return SetDSerror("DirectSoundCaptureCreate8", result);
+        }
+    } else {
+        result = pDirectSoundCreate8(guid, &this->hidden->sound, NULL);
+        if (result != DS_OK) {
+            return SetDSerror("DirectSoundCreate8", result);
+        }
+        result = IDirectSound_SetCooperativeLevel(this->hidden->sound,
+                                                  GetDesktopWindow(),
+                                                  DSSCL_NORMAL);
+        if (result != DS_OK) {
+            return SetDSerror("DirectSound SetCooperativeLevel", result);
+        }
+    }
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_U8:
+        case AUDIO_S16:
+        case AUDIO_S32:
+        case AUDIO_F32:
+            tried_format = SDL_TRUE;
+
+            this->spec.format = test_format;
+
+            /* Update the fragment size as size in bytes */
+            SDL_CalculateAudioSpec(&this->spec);
+
+            bufsize = numchunks * this->spec.size;
+            if ((bufsize < DSBSIZE_MIN) || (bufsize > DSBSIZE_MAX)) {
+                SDL_SetError("Sound buffer size must be between %d and %d",
+                             (int)((DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks),
+                             (int)(DSBSIZE_MAX / numchunks));
+            } else {
+                int rc;
+                WAVEFORMATEXTENSIBLE wfmt;
+                SDL_zero(wfmt);
+                if (this->spec.channels > 2) {
+                    wfmt.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+                    wfmt.Format.cbSize = sizeof(wfmt) - sizeof(WAVEFORMATEX);
+
+                    if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
+                        SDL_memcpy(&wfmt.SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, sizeof(GUID));
+                    } else {
+                        SDL_memcpy(&wfmt.SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID));
+                    }
+                    wfmt.Samples.wValidBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
+
+                    switch (this->spec.channels) {
+                    case 3: /* 3.0 (or 2.1) */
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER;
+                        break;
+                    case 4: /* 4.0 */
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT;
+                        break;
+                    case 5: /* 5.0 (or 4.1) */
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT;
+                        break;
+                    case 6: /* 5.1 */
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT;
+                        break;
+                    case 7: /* 6.1 */
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_BACK_CENTER;
+                        break;
+                    case 8: /* 7.1 */
+                        wfmt.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT;
+                        break;
+                    default:
+                        SDL_assert(0 && "Unsupported channel count!");
+                        break;
+                    }
+                } else if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
+                    wfmt.Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
+                } else {
+                    wfmt.Format.wFormatTag = WAVE_FORMAT_PCM;
+                }
+
+                wfmt.Format.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
+                wfmt.Format.nChannels = this->spec.channels;
+                wfmt.Format.nSamplesPerSec = this->spec.freq;
+                wfmt.Format.nBlockAlign = wfmt.Format.nChannels * (wfmt.Format.wBitsPerSample / 8);
+                wfmt.Format.nAvgBytesPerSec = wfmt.Format.nSamplesPerSec * wfmt.Format.nBlockAlign;
+
+                rc = iscapture ? CreateCaptureBuffer(this, bufsize, (WAVEFORMATEX *)&wfmt) : CreateSecondary(this, bufsize, (WAVEFORMATEX *)&wfmt);
+                if (rc == 0) {
+                    this->hidden->num_buffers = numchunks;
+                    break;
+                }
+            }
+            continue;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        if (tried_format) {
+            return -1; /* CreateSecondary() should have called SDL_SetError(). */
+        }
+        return SDL_SetError("%s: Unsupported audio format", "directsound");
+    }
+
+    /* Playback buffers will auto-start playing in DSOUND_WaitDevice() */
+
+    return 0; /* good to go. */
+}
+
+static void DSOUND_Deinitialize(void)
+{
+#if HAVE_MMDEVICEAPI_H
+    if (SupportsIMMDevice) {
+        SDL_IMMDevice_Quit();
+        SupportsIMMDevice = SDL_FALSE;
+    }
+#endif /* HAVE_MMDEVICEAPI_H */
+    DSOUND_Unload();
+}
+
+static SDL_bool DSOUND_Init(SDL_AudioDriverImpl *impl)
+{
+    if (!DSOUND_Load()) {
+        return SDL_FALSE;
+    }
+
+#if HAVE_MMDEVICEAPI_H
+    SupportsIMMDevice = !(SDL_IMMDevice_Init() < 0);
+#endif /* HAVE_MMDEVICEAPI_H */
+
+    /* Set the function pointers */
+    impl->DetectDevices = DSOUND_DetectDevices;
+    impl->OpenDevice = DSOUND_OpenDevice;
+    impl->PlayDevice = DSOUND_PlayDevice;
+    impl->WaitDevice = DSOUND_WaitDevice;
+    impl->GetDeviceBuf = DSOUND_GetDeviceBuf;
+    impl->CaptureFromDevice = DSOUND_CaptureFromDevice;
+    impl->FlushCapture = DSOUND_FlushCapture;
+    impl->CloseDevice = DSOUND_CloseDevice;
+    impl->FreeDeviceHandle = DSOUND_FreeDeviceHandle;
+    impl->Deinitialize = DSOUND_Deinitialize;
+    impl->GetDefaultAudioInfo = DSOUND_GetDefaultAudioInfo;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->SupportsNonPow2Samples = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap DSOUND_bootstrap = {
+    "directsound", "DirectSound", DSOUND_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_DSOUND */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/directsound/SDL_directsound.h

@@ -0,0 +1,47 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_directsound_h_
+#define SDL_directsound_h_
+
+#include "../../core/windows/SDL_directx.h"
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+/* The DirectSound objects */
+struct SDL_PrivateAudioData
+{
+    LPDIRECTSOUND sound;
+    LPDIRECTSOUNDBUFFER mixbuf;
+    LPDIRECTSOUNDCAPTURE capture;
+    LPDIRECTSOUNDCAPTUREBUFFER capturebuf;
+    int num_buffers;
+    DWORD lastchunk;
+    Uint8 *locked_buf;
+};
+
+#endif /* SDL_directsound_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/disk/SDL_diskaudio.c

@@ -0,0 +1,197 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_DISK
+
+/* Output raw audio data to a file. */
+
+#if HAVE_STDIO_H
+#include <stdio.h>
+#endif
+
+#include "SDL_rwops.h"
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_diskaudio.h"
+
+/* !!! FIXME: these should be SDL hints, not environment variables. */
+/* environment variables and defaults. */
+#define DISKENVR_OUTFILE    "SDL_DISKAUDIOFILE"
+#define DISKDEFAULT_OUTFILE "sdlaudio.raw"
+#define DISKENVR_INFILE     "SDL_DISKAUDIOFILEIN"
+#define DISKDEFAULT_INFILE  "sdlaudio-in.raw"
+#define DISKENVR_IODELAY    "SDL_DISKAUDIODELAY"
+
+/* This function waits until it is possible to write a full sound buffer */
+static void DISKAUDIO_WaitDevice(_THIS)
+{
+    SDL_Delay(_this->hidden->io_delay);
+}
+
+static void DISKAUDIO_PlayDevice(_THIS)
+{
+    const size_t written = SDL_RWwrite(_this->hidden->io,
+                                       _this->hidden->mixbuf,
+                                       1, _this->spec.size);
+
+    /* If we couldn't write, assume fatal error for now */
+    if (written != _this->spec.size) {
+        SDL_OpenedAudioDeviceDisconnected(_this);
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+}
+
+static Uint8 *DISKAUDIO_GetDeviceBuf(_THIS)
+{
+    return _this->hidden->mixbuf;
+}
+
+static int DISKAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = _this->hidden;
+    const int origbuflen = buflen;
+
+    SDL_Delay(h->io_delay);
+
+    if (h->io) {
+        const size_t br = SDL_RWread(h->io, buffer, 1, buflen);
+        buflen -= (int)br;
+        buffer = ((Uint8 *)buffer) + br;
+        if (buflen > 0) { /* EOF (or error, but whatever). */
+            SDL_RWclose(h->io);
+            h->io = NULL;
+        }
+    }
+
+    /* if we ran out of file, just write silence. */
+    SDL_memset(buffer, _this->spec.silence, buflen);
+
+    return origbuflen;
+}
+
+static void DISKAUDIO_FlushCapture(_THIS)
+{
+    /* no op...we don't advance the file pointer or anything. */
+}
+
+static void DISKAUDIO_CloseDevice(_THIS)
+{
+    if (_this->hidden->io != NULL) {
+        SDL_RWclose(_this->hidden->io);
+    }
+    SDL_free(_this->hidden->mixbuf);
+    SDL_free(_this->hidden);
+}
+
+static const char *get_filename(const SDL_bool iscapture, const char *devname)
+{
+    if (devname == NULL) {
+        devname = SDL_getenv(iscapture ? DISKENVR_INFILE : DISKENVR_OUTFILE);
+        if (devname == NULL) {
+            devname = iscapture ? DISKDEFAULT_INFILE : DISKDEFAULT_OUTFILE;
+        }
+    }
+    return devname;
+}
+
+static int DISKAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    void *handle = _this->handle;
+    /* handle != NULL means "user specified the placeholder name on the fake detected device list" */
+    SDL_bool iscapture = _this->iscapture;
+    const char *fname = get_filename(iscapture, handle ? NULL : devname);
+    const char *envr = SDL_getenv(DISKENVR_IODELAY);
+
+    _this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc(sizeof(*_this->hidden));
+    if (_this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(_this->hidden);
+
+    if (envr != NULL) {
+        _this->hidden->io_delay = SDL_atoi(envr);
+    } else {
+        _this->hidden->io_delay = ((_this->spec.samples * 1000) / _this->spec.freq);
+    }
+
+    /* Open the audio device */
+    _this->hidden->io = SDL_RWFromFile(fname, iscapture ? "rb" : "wb");
+    if (_this->hidden->io == NULL) {
+        return -1;
+    }
+
+    /* Allocate mixing buffer */
+    if (!iscapture) {
+        _this->hidden->mixbuf = (Uint8 *)SDL_malloc(_this->spec.size);
+        if (_this->hidden->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(_this->hidden->mixbuf, _this->spec.silence, _this->spec.size);
+    }
+
+    SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO,
+                    "You are using the SDL disk i/o audio driver!\n");
+    SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO,
+                    " %s file [%s].\n", iscapture ? "Reading from" : "Writing to",
+                    fname);
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static void DISKAUDIO_DetectDevices(void)
+{
+    SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *)0x1);
+    SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *)0x2);
+}
+
+static SDL_bool DISKAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->OpenDevice = DISKAUDIO_OpenDevice;
+    impl->WaitDevice = DISKAUDIO_WaitDevice;
+    impl->PlayDevice = DISKAUDIO_PlayDevice;
+    impl->GetDeviceBuf = DISKAUDIO_GetDeviceBuf;
+    impl->CaptureFromDevice = DISKAUDIO_CaptureFromDevice;
+    impl->FlushCapture = DISKAUDIO_FlushCapture;
+
+    impl->CloseDevice = DISKAUDIO_CloseDevice;
+    impl->DetectDevices = DISKAUDIO_DetectDevices;
+
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->SupportsNonPow2Samples = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap DISKAUDIO_bootstrap = {
+    "disk", "direct-to-disk audio", DISKAUDIO_Init, SDL_TRUE
+};
+
+#endif /* SDL_AUDIO_DRIVER_DISK */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/disk/SDL_diskaudio.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_diskaudio_h_
+#define SDL_diskaudio_h_
+
+#include "SDL_rwops.h"
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *_this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    SDL_RWops *io;
+    Uint32 io_delay;
+    Uint8 *mixbuf;
+};
+
+#endif /* SDL_diskaudio_h_ */
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/dsp/SDL_dspaudio.c

@@ -0,0 +1,318 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_OSS
+
+/* Allow access to a raw mixing buffer */
+
+#include <stdio.h>  /* For perror() */
+#include <string.h> /* For strerror() */
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <signal.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+
+#include <sys/soundcard.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "SDL_dspaudio.h"
+
+static void DSP_DetectDevices(void)
+{
+    SDL_EnumUnixAudioDevices(0, NULL);
+}
+
+static void DSP_CloseDevice(_THIS)
+{
+    if (this->hidden->audio_fd >= 0) {
+        close(this->hidden->audio_fd);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int DSP_OpenDevice(_THIS, const char *devname)
+{
+    SDL_bool iscapture = this->iscapture;
+    const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
+    int format;
+    int value;
+    int frag_spec;
+    SDL_AudioFormat test_format;
+
+    /* We don't care what the devname is...we'll try to open anything. */
+    /*  ...but default to first name in the list... */
+    if (devname == NULL) {
+        devname = SDL_GetAudioDeviceName(0, iscapture);
+        if (devname == NULL) {
+            return SDL_SetError("No such audio device");
+        }
+    }
+
+    /* Make sure fragment size stays a power of 2, or OSS fails. */
+    /* I don't know which of these are actually legal values, though... */
+    if (this->spec.channels > 8) {
+        this->spec.channels = 8;
+    } else if (this->spec.channels > 4) {
+        this->spec.channels = 4;
+    } else if (this->spec.channels > 2) {
+        this->spec.channels = 2;
+    }
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *) SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Open the audio device */
+    this->hidden->audio_fd = open(devname, flags | O_CLOEXEC, 0);
+    if (this->hidden->audio_fd < 0) {
+        return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
+    }
+
+    /* Make the file descriptor use blocking i/o with fcntl() */
+    {
+        long ctlflags;
+        ctlflags = fcntl(this->hidden->audio_fd, F_GETFL);
+        ctlflags &= ~O_NONBLOCK;
+        if (fcntl(this->hidden->audio_fd, F_SETFL, ctlflags) < 0) {
+            return SDL_SetError("Couldn't set audio blocking mode");
+        }
+    }
+
+    /* Get a list of supported hardware formats */
+    if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0) {
+        perror("SNDCTL_DSP_GETFMTS");
+        return SDL_SetError("Couldn't get audio format list");
+    }
+
+    /* Try for a closest match on audio format */
+    format = 0;
+    for (test_format = SDL_FirstAudioFormat(this->spec.format);
+         !format && test_format;) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        switch (test_format) {
+        case AUDIO_U8:
+            if (value & AFMT_U8) {
+                format = AFMT_U8;
+            }
+            break;
+        case AUDIO_S16LSB:
+            if (value & AFMT_S16_LE) {
+                format = AFMT_S16_LE;
+            }
+            break;
+        case AUDIO_S16MSB:
+            if (value & AFMT_S16_BE) {
+                format = AFMT_S16_BE;
+            }
+            break;
+#if 0
+/*
+ * These formats are not used by any real life systems so they are not
+ * needed here.
+ */
+        case AUDIO_S8:
+            if (value & AFMT_S8) {
+                format = AFMT_S8;
+            }
+            break;
+        case AUDIO_U16LSB:
+            if (value & AFMT_U16_LE) {
+                format = AFMT_U16_LE;
+            }
+            break;
+        case AUDIO_U16MSB:
+            if (value & AFMT_U16_BE) {
+                format = AFMT_U16_BE;
+            }
+            break;
+#endif
+        default:
+            format = 0;
+            break;
+        }
+        if (!format) {
+            test_format = SDL_NextAudioFormat();
+        }
+    }
+    if (format == 0) {
+        return SDL_SetError("Couldn't find any hardware audio formats");
+    }
+    this->spec.format = test_format;
+
+    /* Set the audio format */
+    value = format;
+    if ((ioctl(this->hidden->audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
+        (value != format)) {
+        perror("SNDCTL_DSP_SETFMT");
+        return SDL_SetError("Couldn't set audio format");
+    }
+
+    /* Set the number of channels of output */
+    value = this->spec.channels;
+    if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0) {
+        perror("SNDCTL_DSP_CHANNELS");
+        return SDL_SetError("Cannot set the number of channels");
+    }
+    this->spec.channels = value;
+
+    /* Set the DSP frequency */
+    value = this->spec.freq;
+    if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_SPEED, &value) < 0) {
+        perror("SNDCTL_DSP_SPEED");
+        return SDL_SetError("Couldn't set audio frequency");
+    }
+    this->spec.freq = value;
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Determine the power of two of the fragment size */
+    for (frag_spec = 0; (0x01U << frag_spec) < this->spec.size; ++frag_spec) {
+    }
+    if ((0x01U << frag_spec) != this->spec.size) {
+        return SDL_SetError("Fragment size must be a power of two");
+    }
+    frag_spec |= 0x00020000; /* two fragments, for low latency */
+
+    /* Set the audio buffering parameters */
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Requesting %d fragments of size %d\n",
+            (frag_spec >> 16), 1 << (frag_spec & 0xFFFF));
+#endif
+    if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0) {
+        perror("SNDCTL_DSP_SETFRAGMENT");
+    }
+#ifdef DEBUG_AUDIO
+    {
+        audio_buf_info info;
+        ioctl(this->hidden->audio_fd, SNDCTL_DSP_GETOSPACE, &info);
+        fprintf(stderr, "fragments = %d\n", info.fragments);
+        fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
+        fprintf(stderr, "fragsize = %d\n", info.fragsize);
+        fprintf(stderr, "bytes = %d\n", info.bytes);
+    }
+#endif
+
+    /* Allocate mixing buffer */
+    if (!iscapture) {
+        this->hidden->mixlen = this->spec.size;
+        this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
+        if (this->hidden->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+    }
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static void DSP_PlayDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    if (write(h->audio_fd, h->mixbuf, h->mixlen) == -1) {
+        perror("Audio write");
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", h->mixlen);
+#endif
+}
+
+static Uint8 *DSP_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static int DSP_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    return (int)read(this->hidden->audio_fd, buffer, buflen);
+}
+
+static void DSP_FlushCapture(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    audio_buf_info info;
+    if (ioctl(h->audio_fd, SNDCTL_DSP_GETISPACE, &info) == 0) {
+        while (info.bytes > 0) {
+            char buf[512];
+            const size_t len = SDL_min(sizeof(buf), info.bytes);
+            const ssize_t br = read(h->audio_fd, buf, len);
+            if (br <= 0) {
+                break;
+            }
+            info.bytes -= br;
+        }
+    }
+}
+
+static SDL_bool InitTimeDevicesExist = SDL_FALSE;
+static int look_for_devices_test(int fd)
+{
+    InitTimeDevicesExist = SDL_TRUE; /* note that _something_ exists. */
+    /* Don't add to the device list, we're just seeing if any devices exist. */
+    return 0;
+}
+
+static SDL_bool DSP_Init(SDL_AudioDriverImpl *impl)
+{
+    InitTimeDevicesExist = SDL_FALSE;
+    SDL_EnumUnixAudioDevices(0, look_for_devices_test);
+    if (!InitTimeDevicesExist) {
+        SDL_SetError("dsp: No such audio device");
+        return SDL_FALSE; /* maybe try a different backend. */
+    }
+
+    /* Set the function pointers */
+    impl->DetectDevices = DSP_DetectDevices;
+    impl->OpenDevice = DSP_OpenDevice;
+    impl->PlayDevice = DSP_PlayDevice;
+    impl->GetDeviceBuf = DSP_GetDeviceBuf;
+    impl->CloseDevice = DSP_CloseDevice;
+    impl->CaptureFromDevice = DSP_CaptureFromDevice;
+    impl->FlushCapture = DSP_FlushCapture;
+
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap DSP_bootstrap = {
+    "dsp", "OSS /dev/dsp standard audio", DSP_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_OSS */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/dsp/SDL_dspaudio.h

@@ -0,0 +1,43 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_dspaudio_h_
+#define SDL_dspaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    int audio_fd;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+};
+#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
+
+#endif /* SDL_dspaudio_h_ */
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/dummy/SDL_dummyaudio.c

@@ -0,0 +1,64 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+/* Output audio to nowhere... */
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_dummyaudio.h"
+
+static int DUMMYAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    _this->hidden = (void *)0x1; /* just something non-NULL */
+
+    return 0; /* always succeeds. */
+}
+
+static int DUMMYAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    /* Delay to make this sort of simulate real audio input. */
+    SDL_Delay((_this->spec.samples * 1000) / _this->spec.freq);
+
+    /* always return a full buffer of silence. */
+    SDL_memset(buffer, _this->spec.silence, buflen);
+    return buflen;
+}
+
+static SDL_bool DUMMYAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->OpenDevice = DUMMYAUDIO_OpenDevice;
+    impl->CaptureFromDevice = DUMMYAUDIO_CaptureFromDevice;
+
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap DUMMYAUDIO_bootstrap = {
+    "dummy", "SDL dummy audio driver", DUMMYAUDIO_Init, SDL_TRUE
+};
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/dummy/SDL_dummyaudio.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_dummyaudio_h_
+#define SDL_dummyaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *_this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    Uint8 *mixbuf;
+    Uint32 mixlen;
+    Uint32 write_delay;
+    Uint32 initial_calls;
+};
+
+#endif /* SDL_dummyaudio_h_ */
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/emscripten/SDL_emscriptenaudio.c

@@ -0,0 +1,410 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_EMSCRIPTEN
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_emscriptenaudio.h"
+
+#include <emscripten/emscripten.h>
+
+/* !!! FIXME: this currently expects that the audio callback runs in the main thread,
+   !!! FIXME:  in intervals when the application isn't running, but that may not be
+   !!! FIXME:  true always once pthread support becomes widespread. Revisit this code
+   !!! FIXME:  at some point and see what needs to be done for that! */
+
+static void FeedAudioDevice(_THIS, const void *buf, const int buflen)
+{
+    const int framelen = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
+    /* *INDENT-OFF* */ /* clang-format off */
+    MAIN_THREAD_EM_ASM({
+        var SDL2 = Module['SDL2'];
+        var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
+        for (var c = 0; c < numChannels; ++c) {
+            var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
+            if (channelData.length != $1) {
+                throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
+            }
+
+            for (var j = 0; j < $1; ++j) {
+                channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2];  /* !!! FIXME: why are these shifts here? */
+            }
+        }
+    }, buf, buflen / framelen);
+/* *INDENT-ON* */ /* clang-format on */
+}
+
+static void HandleAudioProcess(_THIS)
+{
+    SDL_AudioCallback callback = this->callbackspec.callback;
+    const int stream_len = this->callbackspec.size;
+
+    /* Only do something if audio is enabled */
+    if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
+        if (this->stream) {
+            SDL_AudioStreamClear(this->stream);
+        }
+
+        SDL_memset(this->work_buffer, this->spec.silence, this->spec.size);
+        FeedAudioDevice(this, this->work_buffer, this->spec.size);
+        return;
+    }
+
+    if (this->stream == NULL) { /* no conversion necessary. */
+        SDL_assert(this->spec.size == stream_len);
+        callback(this->callbackspec.userdata, this->work_buffer, stream_len);
+    } else { /* streaming/converting */
+        int got;
+        while (SDL_AudioStreamAvailable(this->stream) < ((int)this->spec.size)) {
+            callback(this->callbackspec.userdata, this->work_buffer, stream_len);
+            if (SDL_AudioStreamPut(this->stream, this->work_buffer, stream_len) == -1) {
+                SDL_AudioStreamClear(this->stream);
+                SDL_AtomicSet(&this->enabled, 0);
+                break;
+            }
+        }
+
+        got = SDL_AudioStreamGet(this->stream, this->work_buffer, this->spec.size);
+        SDL_assert((got < 0) || (got == this->spec.size));
+        if (got != this->spec.size) {
+            SDL_memset(this->work_buffer, this->spec.silence, this->spec.size);
+        }
+    }
+
+    FeedAudioDevice(this, this->work_buffer, this->spec.size);
+}
+
+static void HandleCaptureProcess(_THIS)
+{
+    SDL_AudioCallback callback = this->callbackspec.callback;
+    const int stream_len = this->callbackspec.size;
+
+    /* Only do something if audio is enabled */
+    if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
+        SDL_AudioStreamClear(this->stream);
+        return;
+    }
+
+    /* *INDENT-OFF* */ /* clang-format off */
+    MAIN_THREAD_EM_ASM({
+        var SDL2 = Module['SDL2'];
+        var numChannels = SDL2.capture.currentCaptureBuffer.numberOfChannels;
+        for (var c = 0; c < numChannels; ++c) {
+            var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
+            if (channelData.length != $1) {
+                throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
+            }
+
+            if (numChannels == 1) {  /* fastpath this a little for the common (mono) case. */
+                for (var j = 0; j < $1; ++j) {
+                    setValue($0 + (j * 4), channelData[j], 'float');
+                }
+            } else {
+                for (var j = 0; j < $1; ++j) {
+                    setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
+                }
+            }
+        }
+    }, this->work_buffer, (this->spec.size / sizeof(float)) / this->spec.channels);
+/* *INDENT-ON* */ /* clang-format on */
+
+    /* okay, we've got an interleaved float32 array in C now. */
+
+    if (this->stream == NULL) { /* no conversion necessary. */
+        SDL_assert(this->spec.size == stream_len);
+        callback(this->callbackspec.userdata, this->work_buffer, stream_len);
+    } else { /* streaming/converting */
+        if (SDL_AudioStreamPut(this->stream, this->work_buffer, this->spec.size) == -1) {
+            SDL_AtomicSet(&this->enabled, 0);
+        }
+
+        while (SDL_AudioStreamAvailable(this->stream) >= stream_len) {
+            const int got = SDL_AudioStreamGet(this->stream, this->work_buffer, stream_len);
+            SDL_assert((got < 0) || (got == stream_len));
+            if (got != stream_len) {
+                SDL_memset(this->work_buffer, this->callbackspec.silence, stream_len);
+            }
+            callback(this->callbackspec.userdata, this->work_buffer, stream_len); /* Send it to the app. */
+        }
+    }
+}
+
+static void EMSCRIPTENAUDIO_CloseDevice(_THIS)
+{
+    /* *INDENT-OFF* */ /* clang-format off */
+    MAIN_THREAD_EM_ASM({
+        var SDL2 = Module['SDL2'];
+        if ($0) {
+            if (SDL2.capture.silenceTimer !== undefined) {
+                clearTimeout(SDL2.capture.silenceTimer);
+            }
+            if (SDL2.capture.stream !== undefined) {
+                var tracks = SDL2.capture.stream.getAudioTracks();
+                for (var i = 0; i < tracks.length; i++) {
+                    SDL2.capture.stream.removeTrack(tracks[i]);
+                }
+                SDL2.capture.stream = undefined;
+            }
+            if (SDL2.capture.scriptProcessorNode !== undefined) {
+                SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {};
+                SDL2.capture.scriptProcessorNode.disconnect();
+                SDL2.capture.scriptProcessorNode = undefined;
+            }
+            if (SDL2.capture.mediaStreamNode !== undefined) {
+                SDL2.capture.mediaStreamNode.disconnect();
+                SDL2.capture.mediaStreamNode = undefined;
+            }
+            if (SDL2.capture.silenceBuffer !== undefined) {
+                SDL2.capture.silenceBuffer = undefined
+            }
+            SDL2.capture = undefined;
+        } else {
+            if (SDL2.audio.scriptProcessorNode != undefined) {
+                SDL2.audio.scriptProcessorNode.disconnect();
+                SDL2.audio.scriptProcessorNode = undefined;
+            }
+            SDL2.audio = undefined;
+        }
+        if ((SDL2.audioContext !== undefined) && (SDL2.audio === undefined) && (SDL2.capture === undefined)) {
+            SDL2.audioContext.close();
+            SDL2.audioContext = undefined;
+        }
+    }, this->iscapture);
+/* *INDENT-ON* */ /* clang-format on */
+
+#if 0  /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
+    SDL_free(this->hidden);
+#endif
+}
+
+EM_JS_DEPS(sdlaudio, "$autoResumeAudioContext,$dynCall");
+
+static int EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    SDL_AudioFormat test_format;
+    SDL_bool iscapture = this->iscapture;
+    int result;
+
+    /* based on parts of library_sdl.js */
+
+    /* *INDENT-OFF* */ /* clang-format off */
+    /* create context */
+    result = MAIN_THREAD_EM_ASM_INT({
+        if(typeof(Module['SDL2']) === 'undefined') {
+            Module['SDL2'] = {};
+        }
+        var SDL2 = Module['SDL2'];
+        if (!$0) {
+            SDL2.audio = {};
+        } else {
+            SDL2.capture = {};
+        }
+
+        if (!SDL2.audioContext) {
+            if (typeof(AudioContext) !== 'undefined') {
+                SDL2.audioContext = new AudioContext();
+            } else if (typeof(webkitAudioContext) !== 'undefined') {
+                SDL2.audioContext = new webkitAudioContext();
+            }
+            if (SDL2.audioContext) {
+                autoResumeAudioContext(SDL2.audioContext);
+            }
+        }
+        return SDL2.audioContext === undefined ? -1 : 0;
+    }, iscapture);
+/* *INDENT-ON* */ /* clang-format on */
+
+    if (result < 0) {
+        return SDL_SetError("Web Audio API is not available!");
+    }
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_F32: /* web audio only supports floats */
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        /* Didn't find a compatible format :( */
+        return SDL_SetError("%s: Unsupported audio format", "emscripten");
+    }
+    this->spec.format = test_format;
+
+    /* Initialize all variables that we clean on shutdown */
+#if 0  /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
+    this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+#endif
+    this->hidden = (struct SDL_PrivateAudioData *)0x1;
+
+    /* limit to native freq */
+    this->spec.freq = EM_ASM_INT({
+      var SDL2 = Module['SDL2'];
+      return SDL2.audioContext.sampleRate;
+    });
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* *INDENT-OFF* */ /* clang-format off */
+    if (iscapture) {
+        /* The idea is to take the capture media stream, hook it up to an
+           audio graph where we can pass it through a ScriptProcessorNode
+           to access the raw PCM samples and push them to the SDL app's
+           callback. From there, we "process" the audio data into silence
+           and forget about it. */
+
+        /* This should, strictly speaking, use MediaRecorder for capture, but
+           this API is cleaner to use and better supported, and fires a
+           callback whenever there's enough data to fire down into the app.
+           The downside is that we are spending CPU time silencing a buffer
+           that the audiocontext uselessly mixes into any output. On the
+           upside, both of those things are not only run in native code in
+           the browser, they're probably SIMD code, too. MediaRecorder
+           feels like it's a pretty inefficient tapdance in similar ways,
+           to be honest. */
+
+        MAIN_THREAD_EM_ASM({
+            var SDL2 = Module['SDL2'];
+            var have_microphone = function(stream) {
+                //console.log('SDL audio capture: we have a microphone! Replacing silence callback.');
+                if (SDL2.capture.silenceTimer !== undefined) {
+                    clearTimeout(SDL2.capture.silenceTimer);
+                    SDL2.capture.silenceTimer = undefined;
+                }
+                SDL2.capture.mediaStreamNode = SDL2.audioContext.createMediaStreamSource(stream);
+                SDL2.capture.scriptProcessorNode = SDL2.audioContext.createScriptProcessor($1, $0, 1);
+                SDL2.capture.scriptProcessorNode.onaudioprocess = function(audioProcessingEvent) {
+                    if ((SDL2 === undefined) || (SDL2.capture === undefined)) { return; }
+                    audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
+                    SDL2.capture.currentCaptureBuffer = audioProcessingEvent.inputBuffer;
+                    dynCall('vi', $2, [$3]);
+                };
+                SDL2.capture.mediaStreamNode.connect(SDL2.capture.scriptProcessorNode);
+                SDL2.capture.scriptProcessorNode.connect(SDL2.audioContext.destination);
+                SDL2.capture.stream = stream;
+            };
+
+            var no_microphone = function(error) {
+                //console.log('SDL audio capture: we DO NOT have a microphone! (' + error.name + ')...leaving silence callback running.');
+            };
+
+            /* we write silence to the audio callback until the microphone is available (user approves use, etc). */
+            SDL2.capture.silenceBuffer = SDL2.audioContext.createBuffer($0, $1, SDL2.audioContext.sampleRate);
+            SDL2.capture.silenceBuffer.getChannelData(0).fill(0.0);
+            var silence_callback = function() {
+                SDL2.capture.currentCaptureBuffer = SDL2.capture.silenceBuffer;
+                dynCall('vi', $2, [$3]);
+            };
+
+            SDL2.capture.silenceTimer = setTimeout(silence_callback, ($1 / SDL2.audioContext.sampleRate) * 1000);
+
+            if ((navigator.mediaDevices !== undefined) && (navigator.mediaDevices.getUserMedia !== undefined)) {
+                navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(have_microphone).catch(no_microphone);
+            } else if (navigator.webkitGetUserMedia !== undefined) {
+                navigator.webkitGetUserMedia({ audio: true, video: false }, have_microphone, no_microphone);
+            }
+        }, this->spec.channels, this->spec.samples, HandleCaptureProcess, this);
+    } else {
+        /* setup a ScriptProcessorNode */
+        MAIN_THREAD_EM_ASM({
+            var SDL2 = Module['SDL2'];
+            SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
+            SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
+                if ((SDL2 === undefined) || (SDL2.audio === undefined)) { return; }
+                SDL2.audio.currentOutputBuffer = e['outputBuffer'];
+                dynCall('vi', $2, [$3]);
+            };
+            SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
+        }, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
+    }
+/* *INDENT-ON* */ /* clang-format on */
+
+    return 0;
+}
+
+static void EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice *device)
+{
+}
+
+static SDL_bool EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    SDL_bool available, capture_available;
+
+    /* Set the function pointers */
+    impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
+    impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;
+
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+
+    /* no threads here */
+    impl->LockDevice = impl->UnlockDevice = EMSCRIPTENAUDIO_LockOrUnlockDeviceWithNoMixerLock;
+    impl->ProvidesOwnCallbackThread = SDL_TRUE;
+
+    /* *INDENT-OFF* */ /* clang-format off */
+    /* check availability */
+    available = MAIN_THREAD_EM_ASM_INT({
+        if (typeof(AudioContext) !== 'undefined') {
+            return true;
+        } else if (typeof(webkitAudioContext) !== 'undefined') {
+            return true;
+        }
+        return false;
+    });
+/* *INDENT-ON* */ /* clang-format on */
+
+    if (!available) {
+        SDL_SetError("No audio context available");
+    }
+
+    /* *INDENT-OFF* */ /* clang-format off */
+    capture_available = available && MAIN_THREAD_EM_ASM_INT({
+        if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
+            return true;
+        } else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
+            return true;
+        }
+        return false;
+    });
+/* *INDENT-ON* */ /* clang-format on */
+
+    impl->HasCaptureSupport = capture_available ? SDL_TRUE : SDL_FALSE;
+    impl->OnlyHasDefaultCaptureDevice = capture_available ? SDL_TRUE : SDL_FALSE;
+
+    return available;
+}
+
+AudioBootStrap EMSCRIPTENAUDIO_bootstrap = {
+    "emscripten", "SDL emscripten audio driver", EMSCRIPTENAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/emscripten/SDL_emscriptenaudio.h

@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_emscriptenaudio_h_
+#define SDL_emscriptenaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    int unused;
+};
+
+#endif /* SDL_emscriptenaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/esd/SDL_esdaudio.c

@@ -0,0 +1,322 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_ESD
+
+/* Allow access to an ESD network stream mixing buffer */
+
+#include <sys/types.h>
+#include <unistd.h>
+#include <signal.h>
+#include <errno.h>
+#include <esd.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_esdaudio.h"
+
+#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC
+#include "SDL_name.h"
+#include "SDL_loadso.h"
+#else
+#define SDL_NAME(X) X
+#endif
+
+#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC
+
+static const char *esd_library = SDL_AUDIO_DRIVER_ESD_DYNAMIC;
+static void *esd_handle = NULL;
+
+static int (*SDL_NAME(esd_open_sound)) (const char *host);
+static int (*SDL_NAME(esd_close)) (int esd);
+static int (*SDL_NAME(esd_play_stream)) (esd_format_t format, int rate,
+                                         const char *host, const char *name);
+
+#define SDL_ESD_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
+static struct
+{
+    const char *name;
+    void **func;
+} const esd_functions[] = {
+    SDL_ESD_SYM(esd_open_sound),
+    SDL_ESD_SYM(esd_close), SDL_ESD_SYM(esd_play_stream),
+};
+
+#undef SDL_ESD_SYM
+
+static void UnloadESDLibrary()
+{
+    if (esd_handle != NULL) {
+        SDL_UnloadObject(esd_handle);
+        esd_handle = NULL;
+    }
+}
+
+static int LoadESDLibrary(void)
+{
+    int i, retval = -1;
+
+    if (esd_handle == NULL) {
+        esd_handle = SDL_LoadObject(esd_library);
+        if (esd_handle) {
+            retval = 0;
+            for (i = 0; i < SDL_arraysize(esd_functions); ++i) {
+                *esd_functions[i].func =
+                    SDL_LoadFunction(esd_handle, esd_functions[i].name);
+                if (!*esd_functions[i].func) {
+                    retval = -1;
+                    UnloadESDLibrary();
+                    break;
+                }
+            }
+        }
+    }
+    return retval;
+}
+
+#else
+
+static void UnloadESDLibrary()
+{
+    return;
+}
+
+static int LoadESDLibrary(void)
+{
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_ESD_DYNAMIC */
+
+
+/* This function waits until it is possible to write a full sound buffer */
+static void ESD_WaitDevice(_THIS)
+{
+    Sint32 ticks;
+
+    /* Check to see if the thread-parent process is still alive */
+    {
+        static int cnt = 0;
+        /* Note that this only works with thread implementations
+           that use a different process id for each thread.
+         */
+        /* Check every 10 loops */
+        if (this->hidden->parent && (((++cnt) % 10) == 0)) {
+            if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
+                SDL_OpenedAudioDeviceDisconnected(this);
+            }
+        }
+    }
+
+    /* Use timer for general audio synchronization */
+    ticks = ((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
+    if (ticks > 0) {
+        SDL_Delay(ticks);
+    }
+}
+
+static void ESD_PlayDevice(_THIS)
+{
+    int written = 0;
+
+    /* Write the audio data, checking for EAGAIN on broken audio drivers */
+    do {
+        written = write(this->hidden->audio_fd,
+                        this->hidden->mixbuf, this->hidden->mixlen);
+        if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
+            SDL_Delay(1);       /* Let a little CPU time go by */
+        }
+    } while ((written < 0) &&
+             ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));
+
+    /* Set the next write frame */
+    this->hidden->next_frame += this->hidden->frame_ticks;
+
+    /* If we couldn't write, assume fatal error for now */
+    if (written < 0) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+}
+
+static Uint8 *ESD_GetDeviceBuf(_THIS)
+{
+    return (this->hidden->mixbuf);
+}
+
+static void ESD_CloseDevice(_THIS)
+{
+    if (this->hidden->audio_fd >= 0) {
+        SDL_NAME(esd_close) (this->hidden->audio_fd);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+/* Try to get the name of the program */
+static char *get_progname(void)
+{
+    char *progname = NULL;
+#ifdef __LINUX__
+    FILE *fp;
+    static char temp[BUFSIZ];
+
+    SDL_snprintf(temp, SDL_arraysize(temp), "/proc/%d/cmdline", getpid());
+    fp = fopen(temp, "r");
+    if (fp != NULL) {
+        if (fgets(temp, sizeof(temp) - 1, fp)) {
+            progname = SDL_strrchr(temp, '/');
+            if (progname == NULL) {
+                progname = temp;
+            } else {
+                progname = progname + 1;
+            }
+        }
+        fclose(fp);
+    }
+#endif
+    return (progname);
+}
+
+
+static int ESD_OpenDevice(_THIS, const char *devname)
+{
+    esd_format_t format = (ESD_STREAM | ESD_PLAY);
+    SDL_AudioFormat test_format = 0;
+    int found = 0;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+    this->hidden->audio_fd = -1;
+
+    /* Convert audio spec to the ESD audio format */
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format);
+         !found && test_format; test_format = SDL_NextAudioFormat()) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        found = 1;
+        switch (test_format) {
+        case AUDIO_U8:
+            format |= ESD_BITS8;
+            break;
+        case AUDIO_S16SYS:
+            format |= ESD_BITS16;
+            break;
+        default:
+            found = 0;
+            break;
+        }
+    }
+
+    if (!found) {
+        return SDL_SetError("Couldn't find any hardware audio formats");
+    }
+
+    if (this->spec.channels == 1) {
+        format |= ESD_MONO;
+    } else {
+        format |= ESD_STEREO;
+    }
+#if 0
+    this->spec.samples = ESD_BUF_SIZE;  /* Darn, no way to change this yet */
+#endif
+
+    /* Open a connection to the ESD audio server */
+    this->hidden->audio_fd =
+        SDL_NAME(esd_play_stream) (format, this->spec.freq, NULL,
+                                   get_progname());
+
+    if (this->hidden->audio_fd < 0) {
+        return SDL_SetError("Couldn't open ESD connection");
+    }
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+    this->hidden->frame_ticks =
+        (float) (this->spec.samples * 1000) / this->spec.freq;
+    this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
+
+    /* Allocate mixing buffer */
+    this->hidden->mixlen = this->spec.size;
+    this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+
+    /* Get the parent process id (we're the parent of the audio thread) */
+    this->hidden->parent = getpid();
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static void ESD_Deinitialize(void)
+{
+    UnloadESDLibrary();
+}
+
+static SDL_bool ESD_Init(SDL_AudioDriverImpl * impl)
+{
+    if (LoadESDLibrary() < 0) {
+        return SDL_FALSE;
+    } else {
+        int connection = 0;
+
+        /* Don't start ESD if it's not running */
+        SDL_setenv("ESD_NO_SPAWN", "1", 0);
+
+        connection = SDL_NAME(esd_open_sound) (NULL);
+        if (connection < 0) {
+            UnloadESDLibrary();
+            SDL_SetError("ESD: esd_open_sound failed (no audio server?)");
+            return SDL_FALSE;
+        }
+        SDL_NAME(esd_close) (connection);
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = ESD_OpenDevice;
+    impl->PlayDevice = ESD_PlayDevice;
+    impl->WaitDevice = ESD_WaitDevice;
+    impl->GetDeviceBuf = ESD_GetDeviceBuf;
+    impl->CloseDevice = ESD_CloseDevice;
+    impl->Deinitialize = ESD_Deinitialize;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+
+AudioBootStrap ESD_bootstrap = {
+    "esd", "Enlightened Sound Daemon", ESD_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_ESD */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/esd/SDL_esdaudio.h

@@ -0,0 +1,51 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_esdaudio_h_
+#define SDL_esdaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    int audio_fd;
+
+    /* The parent process id, to detect when application quits */
+    pid_t parent;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    /* Support for audio timing using a timer */
+    float frame_ticks;
+    float next_frame;
+};
+#define FUDGE_TICKS 10      /* The scheduler overhead ticks per frame */
+
+#endif /* SDL_esdaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/fusionsound/SDL_fsaudio.c

@@ -0,0 +1,305 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_FUSIONSOUND
+
+/* !!! FIXME: why is this is SDL_FS_* instead of FUSIONSOUND_*? */
+
+/* Allow access to a raw mixing buffer */
+
+#ifdef HAVE_SIGNAL_H
+#include <signal.h>
+#endif
+#include <unistd.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_fsaudio.h"
+
+#include <fusionsound/fusionsound_version.h>
+
+/* #define SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC "libfusionsound.so" */
+
+#ifdef SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC
+#include "SDL_name.h"
+#include "SDL_loadso.h"
+#else
+#define SDL_NAME(X) X
+#endif
+
+#if (FUSIONSOUND_MAJOR_VERSION == 1) && (FUSIONSOUND_MINOR_VERSION < 1)
+typedef DFBResult DirectResult;
+#endif
+
+/* Buffers to use - more than 2 gives a lot of latency */
+#define FUSION_BUFFERS              (2)
+
+#ifdef SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC
+
+static const char *fs_library = SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC;
+static void *fs_handle = NULL;
+
+static DirectResult (*SDL_NAME(FusionSoundInit)) (int *argc, char *(*argv[]));
+static DirectResult (*SDL_NAME(FusionSoundCreate)) (IFusionSound **
+                                                   ret_interface);
+
+#define SDL_FS_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
+static struct
+{
+    const char *name;
+    void **func;
+} fs_functions[] = {
+/* *INDENT-OFF* */
+    SDL_FS_SYM(FusionSoundInit),
+    SDL_FS_SYM(FusionSoundCreate),
+/* *INDENT-ON* */
+};
+
+#undef SDL_FS_SYM
+
+static void UnloadFusionSoundLibrary()
+{
+    if (fs_handle != NULL) {
+        SDL_UnloadObject(fs_handle);
+        fs_handle = NULL;
+    }
+}
+
+static int LoadFusionSoundLibrary(void)
+{
+    int i, retval = -1;
+
+    if (fs_handle == NULL) {
+        fs_handle = SDL_LoadObject(fs_library);
+        if (fs_handle != NULL) {
+            retval = 0;
+            for (i = 0; i < SDL_arraysize(fs_functions); ++i) {
+                *fs_functions[i].func =
+                    SDL_LoadFunction(fs_handle, fs_functions[i].name);
+                if (!*fs_functions[i].func) {
+                    retval = -1;
+                    UnloadFusionSoundLibrary();
+                    break;
+                }
+            }
+        }
+    }
+
+    return retval;
+}
+
+#else
+
+static void UnloadFusionSoundLibrary()
+{
+    return;
+}
+
+static int LoadFusionSoundLibrary(void)
+{
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC */
+
+/* This function waits until it is possible to write a full sound buffer */
+static void SDL_FS_WaitDevice(_THIS)
+{
+    this->hidden->stream->Wait(this->hidden->stream,
+                               this->hidden->mixsamples);
+}
+
+static void SDL_FS_PlayDevice(_THIS)
+{
+    DirectResult ret;
+
+    ret = this->hidden->stream->Write(this->hidden->stream,
+                                      this->hidden->mixbuf,
+                                      this->hidden->mixsamples);
+    /* If we couldn't write, assume fatal error for now */
+    if (ret) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen);
+#endif
+}
+
+
+static Uint8 *SDL_FS_GetDeviceBuf(_THIS)
+{
+    return (this->hidden->mixbuf);
+}
+
+
+static void SDL_FS_CloseDevice(_THIS)
+{
+    if (this->hidden->stream) {
+        this->hidden->stream->Release(this->hidden->stream);
+    }
+    if (this->hidden->fs) {
+        this->hidden->fs->Release(this->hidden->fs);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+
+static int SDL_FS_OpenDevice(_THIS, const char *devname)
+{
+    int bytes;
+    SDL_AudioFormat test_format;
+    FSSampleFormat fs_format;
+    FSStreamDescription desc;
+    DirectResult ret;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        switch (test_format) {
+        case AUDIO_U8:
+            fs_format = FSSF_U8;
+            break;
+        case AUDIO_S16SYS:
+            fs_format = FSSF_S16;
+            break;
+        case AUDIO_S32SYS:
+            fs_format = FSSF_S32;
+            break;
+        case AUDIO_F32SYS:
+            fs_format = FSSF_FLOAT;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "fusionsound");
+    }
+    this->spec.format = test_format;
+    bytes = SDL_AUDIO_BITSIZE(test_format) / 8;
+
+    /* Retrieve the main sound interface. */
+    ret = SDL_NAME(FusionSoundCreate) (&this->hidden->fs);
+    if (ret) {
+        return SDL_SetError("Unable to initialize FusionSound: %d", ret);
+    }
+
+    this->hidden->mixsamples = this->spec.size / bytes / this->spec.channels;
+
+    /* Fill stream description. */
+    desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
+        FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT | FSSDF_PREBUFFER;
+    desc.samplerate = this->spec.freq;
+    desc.buffersize = this->spec.size * FUSION_BUFFERS;
+    desc.channels = this->spec.channels;
+    desc.prebuffer = 10;
+    desc.sampleformat = fs_format;
+
+    ret =
+        this->hidden->fs->CreateStream(this->hidden->fs, &desc,
+                                       &this->hidden->stream);
+    if (ret) {
+        return SDL_SetError("Unable to create FusionSoundStream: %d", ret);
+    }
+
+    /* See what we got */
+    desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
+        FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT;
+    ret = this->hidden->stream->GetDescription(this->hidden->stream, &desc);
+
+    this->spec.freq = desc.samplerate;
+    this->spec.size =
+        desc.buffersize / FUSION_BUFFERS * bytes * desc.channels;
+    this->spec.channels = desc.channels;
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    this->hidden->mixlen = this->spec.size;
+    this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+
+static void SDL_FS_Deinitialize(void)
+{
+    UnloadFusionSoundLibrary();
+}
+
+
+static SDL_bool SDL_FS_Init(SDL_AudioDriverImpl * impl)
+{
+    if (LoadFusionSoundLibrary() < 0) {
+        return SDL_FALSE;
+    } else {
+        DirectResult ret;
+
+        ret = SDL_NAME(FusionSoundInit) (NULL, NULL);
+        if (ret) {
+            UnloadFusionSoundLibrary();
+            SDL_SetError
+                ("FusionSound: SDL_FS_init failed (FusionSoundInit: %d)",
+                 ret);
+            return SDL_FALSE;
+        }
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = SDL_FS_OpenDevice;
+    impl->PlayDevice = SDL_FS_PlayDevice;
+    impl->WaitDevice = SDL_FS_WaitDevice;
+    impl->GetDeviceBuf = SDL_FS_GetDeviceBuf;
+    impl->CloseDevice = SDL_FS_CloseDevice;
+    impl->Deinitialize = SDL_FS_Deinitialize;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+
+AudioBootStrap FUSIONSOUND_bootstrap = {
+    "fusionsound", "FusionSound", SDL_FS_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_FUSIONSOUND */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/fusionsound/SDL_fsaudio.h

@@ -0,0 +1,50 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_fsaudio_h_
+#define SDL_fsaudio_h_
+
+#include <fusionsound/fusionsound.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* Interface */
+    IFusionSound *fs;
+
+    /* The stream interface for the audio device */
+    IFusionSoundStream *stream;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+    int mixsamples;
+
+};
+
+#endif /* SDL_fsaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/haiku/SDL_haikuaudio.cc

@@ -0,0 +1,238 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_HAIKU
+
+/* Allow access to the audio stream on Haiku */
+
+#include <SoundPlayer.h>
+#include <signal.h>
+
+#include "../../main/haiku/SDL_BeApp.h"
+
+extern "C"
+{
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_haikuaudio.h"
+
+}
+
+
+/* !!! FIXME: have the callback call the higher level to avoid code dupe. */
+/* The Haiku callback for handling the audio buffer */
+static void FillSound(void *device, void *stream, size_t len,
+          const media_raw_audio_format & format)
+{
+    SDL_AudioDevice *audio = (SDL_AudioDevice *) device;
+    SDL_AudioCallback callback = audio->callbackspec.callback;
+
+    SDL_LockMutex(audio->mixer_lock);
+
+    /* Only do something if audio is enabled */
+    if (!SDL_AtomicGet(&audio->enabled) || SDL_AtomicGet(&audio->paused)) {
+        if (audio->stream) {
+            SDL_AudioStreamClear(audio->stream);
+        }
+        SDL_memset(stream, audio->spec.silence, len);
+    } else {
+        SDL_assert(audio->spec.size == len);
+
+        if (audio->stream == NULL) {  /* no conversion necessary. */
+            callback(audio->callbackspec.userdata, (Uint8 *) stream, len);
+        } else {  /* streaming/converting */
+            const int stream_len = audio->callbackspec.size;
+            const int ilen = (int) len;
+            while (SDL_AudioStreamAvailable(audio->stream) < ilen) {
+                callback(audio->callbackspec.userdata, audio->work_buffer, stream_len);
+                if (SDL_AudioStreamPut(audio->stream, audio->work_buffer, stream_len) == -1) {
+                    SDL_AudioStreamClear(audio->stream);
+                    SDL_AtomicSet(&audio->enabled, 0);
+                    break;
+                }
+            }
+
+            const int got = SDL_AudioStreamGet(audio->stream, stream, ilen);
+            SDL_assert((got < 0) || (got == ilen));
+            if (got != ilen) {
+                SDL_memset(stream, audio->spec.silence, len);
+            }
+        }
+    }
+
+    SDL_UnlockMutex(audio->mixer_lock);
+}
+
+static void HAIKUAUDIO_CloseDevice(_THIS)
+{
+    if (_this->hidden->audio_obj) {
+        _this->hidden->audio_obj->Stop();
+        delete _this->hidden->audio_obj;
+    }
+    delete _this->hidden;
+}
+
+
+static const int sig_list[] = {
+    SIGHUP, SIGINT, SIGQUIT, SIGPIPE, SIGALRM, SIGTERM, SIGWINCH, 0
+};
+
+static inline void MaskSignals(sigset_t * omask)
+{
+    sigset_t mask;
+    int i;
+
+    sigemptyset(&mask);
+    for (i = 0; sig_list[i]; ++i) {
+        sigaddset(&mask, sig_list[i]);
+    }
+    sigprocmask(SIG_BLOCK, &mask, omask);
+}
+
+static inline void UnmaskSignals(sigset_t * omask)
+{
+    sigprocmask(SIG_SETMASK, omask, NULL);
+}
+
+
+static int HAIKUAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    media_raw_audio_format format;
+    SDL_AudioFormat test_format;
+
+    /* Initialize all variables that we clean on shutdown */
+    _this->hidden = new SDL_PrivateAudioData;
+    if (_this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(_this->hidden);
+
+    /* Parse the audio format and fill the Be raw audio format */
+    SDL_zero(format);
+    format.byte_order = B_MEDIA_LITTLE_ENDIAN;
+    format.frame_rate = (float) _this->spec.freq;
+    format.channel_count = _this->spec.channels;        /* !!! FIXME: support > 2? */
+    for (test_format = SDL_FirstAudioFormat(_this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_S8:
+            format.format = media_raw_audio_format::B_AUDIO_CHAR;
+            break;
+
+        case AUDIO_U8:
+            format.format = media_raw_audio_format::B_AUDIO_UCHAR;
+            break;
+
+        case AUDIO_S16LSB:
+            format.format = media_raw_audio_format::B_AUDIO_SHORT;
+            break;
+
+        case AUDIO_S16MSB:
+            format.format = media_raw_audio_format::B_AUDIO_SHORT;
+            format.byte_order = B_MEDIA_BIG_ENDIAN;
+            break;
+
+        case AUDIO_S32LSB:
+            format.format = media_raw_audio_format::B_AUDIO_INT;
+            break;
+
+        case AUDIO_S32MSB:
+            format.format = media_raw_audio_format::B_AUDIO_INT;
+            format.byte_order = B_MEDIA_BIG_ENDIAN;
+            break;
+
+        case AUDIO_F32LSB:
+            format.format = media_raw_audio_format::B_AUDIO_FLOAT;
+            break;
+
+        case AUDIO_F32MSB:
+            format.format = media_raw_audio_format::B_AUDIO_FLOAT;
+            format.byte_order = B_MEDIA_BIG_ENDIAN;
+            break;
+
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {      /* shouldn't happen, but just in case... */
+        return SDL_SetError("%s: Unsupported audio format", "haiku");
+    }
+    _this->spec.format = test_format;
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&_this->spec);
+
+    format.buffer_size = _this->spec.size;
+
+    /* Subscribe to the audio stream (creates a new thread) */
+    sigset_t omask;
+    MaskSignals(&omask);
+    _this->hidden->audio_obj = new BSoundPlayer(&format, "SDL Audio",
+                                                FillSound, NULL, _this);
+    UnmaskSignals(&omask);
+
+    if (_this->hidden->audio_obj->Start() == B_NO_ERROR) {
+        _this->hidden->audio_obj->SetHasData(true);
+    } else {
+        return SDL_SetError("Unable to start Be audio");
+    }
+
+    /* We're running! */
+    return 0;
+}
+
+static void HAIKUAUDIO_Deinitialize(void)
+{
+    SDL_QuitBeApp();
+}
+
+static SDL_bool HAIKUAUDIO_Init(SDL_AudioDriverImpl * impl)
+{
+    /* Initialize the Be Application, if it's not already started */
+    if (SDL_InitBeApp() < 0) {
+        return SDL_FALSE;
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = HAIKUAUDIO_OpenDevice;
+    impl->CloseDevice = HAIKUAUDIO_CloseDevice;
+    impl->Deinitialize = HAIKUAUDIO_Deinitialize;
+    impl->ProvidesOwnCallbackThread = SDL_TRUE;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+extern "C"
+{
+    extern AudioBootStrap HAIKUAUDIO_bootstrap;
+}
+AudioBootStrap HAIKUAUDIO_bootstrap = {
+    "haiku", "Haiku BSoundPlayer", HAIKUAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_HAIKU */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/haiku/SDL_haikuaudio.h

@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_haikuaudio_h_
+#define SDL_haikuaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *_this
+
+struct SDL_PrivateAudioData
+{
+    BSoundPlayer *audio_obj;
+};
+
+#endif /* SDL_haikuaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/jack/SDL_jackaudio.c

@@ -0,0 +1,431 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_JACK
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_jackaudio.h"
+#include "SDL_loadso.h"
+#include "../../thread/SDL_systhread.h"
+
+static jack_client_t *(*JACK_jack_client_open)(const char *, jack_options_t, jack_status_t *, ...);
+static int (*JACK_jack_client_close)(jack_client_t *);
+static void (*JACK_jack_on_shutdown)(jack_client_t *, JackShutdownCallback, void *);
+static int (*JACK_jack_activate)(jack_client_t *);
+static int (*JACK_jack_deactivate)(jack_client_t *);
+static void *(*JACK_jack_port_get_buffer)(jack_port_t *, jack_nframes_t);
+static int (*JACK_jack_port_unregister)(jack_client_t *, jack_port_t *);
+static void (*JACK_jack_free)(void *);
+static const char **(*JACK_jack_get_ports)(jack_client_t *, const char *, const char *, unsigned long);
+static jack_nframes_t (*JACK_jack_get_sample_rate)(jack_client_t *);
+static jack_nframes_t (*JACK_jack_get_buffer_size)(jack_client_t *);
+static jack_port_t *(*JACK_jack_port_register)(jack_client_t *, const char *, const char *, unsigned long, unsigned long);
+static jack_port_t *(*JACK_jack_port_by_name)(jack_client_t *, const char *);
+static const char *(*JACK_jack_port_name)(const jack_port_t *);
+static const char *(*JACK_jack_port_type)(const jack_port_t *);
+static int (*JACK_jack_connect)(jack_client_t *, const char *, const char *);
+static int (*JACK_jack_set_process_callback)(jack_client_t *, JackProcessCallback, void *);
+
+static int load_jack_syms(void);
+
+#ifdef SDL_AUDIO_DRIVER_JACK_DYNAMIC
+
+static const char *jack_library = SDL_AUDIO_DRIVER_JACK_DYNAMIC;
+static void *jack_handle = NULL;
+
+/* !!! FIXME: this is copy/pasted in several places now */
+static int load_jack_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(jack_handle, fn);
+    if (*addr == NULL) {
+        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        return 0;
+    }
+
+    return 1;
+}
+
+/* cast funcs to char* first, to please GCC's strict aliasing rules. */
+#define SDL_JACK_SYM(x)                                 \
+    if (!load_jack_sym(#x, (void **)(char *)&JACK_##x)) \
+    return -1
+
+static void UnloadJackLibrary(void)
+{
+    if (jack_handle != NULL) {
+        SDL_UnloadObject(jack_handle);
+        jack_handle = NULL;
+    }
+}
+
+static int LoadJackLibrary(void)
+{
+    int retval = 0;
+    if (jack_handle == NULL) {
+        jack_handle = SDL_LoadObject(jack_library);
+        if (jack_handle == NULL) {
+            retval = -1;
+            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
+        } else {
+            retval = load_jack_syms();
+            if (retval < 0) {
+                UnloadJackLibrary();
+            }
+        }
+    }
+    return retval;
+}
+
+#else
+
+#define SDL_JACK_SYM(x) JACK_##x = x
+
+static void UnloadJackLibrary(void)
+{
+}
+
+static int LoadJackLibrary(void)
+{
+    load_jack_syms();
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_JACK_DYNAMIC */
+
+static int load_jack_syms(void)
+{
+    SDL_JACK_SYM(jack_client_open);
+    SDL_JACK_SYM(jack_client_close);
+    SDL_JACK_SYM(jack_on_shutdown);
+    SDL_JACK_SYM(jack_activate);
+    SDL_JACK_SYM(jack_deactivate);
+    SDL_JACK_SYM(jack_port_get_buffer);
+    SDL_JACK_SYM(jack_port_unregister);
+    SDL_JACK_SYM(jack_free);
+    SDL_JACK_SYM(jack_get_ports);
+    SDL_JACK_SYM(jack_get_sample_rate);
+    SDL_JACK_SYM(jack_get_buffer_size);
+    SDL_JACK_SYM(jack_port_register);
+    SDL_JACK_SYM(jack_port_by_name);
+    SDL_JACK_SYM(jack_port_name);
+    SDL_JACK_SYM(jack_port_type);
+    SDL_JACK_SYM(jack_connect);
+    SDL_JACK_SYM(jack_set_process_callback);
+
+    return 0;
+}
+
+static void jackShutdownCallback(void *arg) /* JACK went away; device is lost. */
+{
+    SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
+    SDL_OpenedAudioDeviceDisconnected(this);
+    SDL_SemPost(this->hidden->iosem); /* unblock the SDL thread. */
+}
+
+// !!! FIXME: implement and register these!
+// typedef int(* JackSampleRateCallback)(jack_nframes_t nframes, void *arg)
+// typedef int(* JackBufferSizeCallback)(jack_nframes_t nframes, void *arg)
+
+static int jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
+{
+    SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
+    jack_port_t **ports = this->hidden->sdlports;
+    const int total_channels = this->spec.channels;
+    const int total_frames = this->spec.samples;
+    int channelsi;
+
+    if (!SDL_AtomicGet(&this->enabled)) {
+        /* silence the buffer to avoid repeats and corruption. */
+        SDL_memset(this->hidden->iobuffer, '\0', this->spec.size);
+    }
+
+    for (channelsi = 0; channelsi < total_channels; channelsi++) {
+        float *dst = (float *)JACK_jack_port_get_buffer(ports[channelsi], nframes);
+        if (dst) {
+            const float *src = this->hidden->iobuffer + channelsi;
+            int framesi;
+            for (framesi = 0; framesi < total_frames; framesi++) {
+                *(dst++) = *src;
+                src += total_channels;
+            }
+        }
+    }
+
+    SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; refill the buffer. */
+    return 0;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void JACK_WaitDevice(_THIS)
+{
+    if (SDL_AtomicGet(&this->enabled)) {
+        if (SDL_SemWait(this->hidden->iosem) == -1) {
+            SDL_OpenedAudioDeviceDisconnected(this);
+        }
+    }
+}
+
+static Uint8 *JACK_GetDeviceBuf(_THIS)
+{
+    return (Uint8 *)this->hidden->iobuffer;
+}
+
+static int jackProcessCaptureCallback(jack_nframes_t nframes, void *arg)
+{
+    SDL_AudioDevice *this = (SDL_AudioDevice *)arg;
+    if (SDL_AtomicGet(&this->enabled)) {
+        jack_port_t **ports = this->hidden->sdlports;
+        const int total_channels = this->spec.channels;
+        const int total_frames = this->spec.samples;
+        int channelsi;
+
+        for (channelsi = 0; channelsi < total_channels; channelsi++) {
+            const float *src = (const float *)JACK_jack_port_get_buffer(ports[channelsi], nframes);
+            if (src) {
+                float *dst = this->hidden->iobuffer + channelsi;
+                int framesi;
+                for (framesi = 0; framesi < total_frames; framesi++) {
+                    *dst = *(src++);
+                    dst += total_channels;
+                }
+            }
+        }
+    }
+
+    SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; new buffer is ready! */
+    return 0;
+}
+
+static int JACK_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    SDL_assert(buflen == this->spec.size); /* we always fill a full buffer. */
+
+    /* Wait for JACK to fill the iobuffer */
+    if (SDL_SemWait(this->hidden->iosem) == -1) {
+        return -1;
+    }
+
+    SDL_memcpy(buffer, this->hidden->iobuffer, buflen);
+    return buflen;
+}
+
+static void JACK_FlushCapture(_THIS)
+{
+    SDL_SemWait(this->hidden->iosem);
+}
+
+static void JACK_CloseDevice(_THIS)
+{
+    if (this->hidden->client) {
+        JACK_jack_deactivate(this->hidden->client);
+
+        if (this->hidden->sdlports) {
+            const int channels = this->spec.channels;
+            int i;
+            for (i = 0; i < channels; i++) {
+                JACK_jack_port_unregister(this->hidden->client, this->hidden->sdlports[i]);
+            }
+            SDL_free(this->hidden->sdlports);
+        }
+
+        JACK_jack_client_close(this->hidden->client);
+    }
+
+    if (this->hidden->iosem) {
+        SDL_DestroySemaphore(this->hidden->iosem);
+    }
+
+    SDL_free(this->hidden->iobuffer);
+    SDL_free(this->hidden);
+}
+
+static int JACK_OpenDevice(_THIS, const char *devname)
+{
+    /* Note that JACK uses "output" for capture devices (they output audio
+        data to us) and "input" for playback (we input audio data to them).
+        Likewise, SDL's playback port will be "output" (we write data out)
+        and capture will be "input" (we read data in). */
+    SDL_bool iscapture = this->iscapture;
+    const unsigned long sysportflags = iscapture ? JackPortIsOutput : JackPortIsInput;
+    const unsigned long sdlportflags = iscapture ? JackPortIsInput : JackPortIsOutput;
+    const JackProcessCallback callback = iscapture ? jackProcessCaptureCallback : jackProcessPlaybackCallback;
+    const char *sdlportstr = iscapture ? "input" : "output";
+    const char **devports = NULL;
+    int *audio_ports;
+    jack_client_t *client = NULL;
+    jack_status_t status;
+    int channels = 0;
+    int ports = 0;
+    int i;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    /* !!! FIXME: we _still_ need an API to specify an app name */
+    client = JACK_jack_client_open("SDL", JackNoStartServer, &status, NULL);
+    this->hidden->client = client;
+    if (client == NULL) {
+        return SDL_SetError("Can't open JACK client");
+    }
+
+    devports = JACK_jack_get_ports(client, NULL, NULL, JackPortIsPhysical | sysportflags);
+    if (devports == NULL || !devports[0]) {
+        return SDL_SetError("No physical JACK ports available");
+    }
+
+    while (devports[++ports]) {
+        /* spin to count devports */
+    }
+
+    /* Filter out non-audio ports */
+    audio_ports = SDL_calloc(ports, sizeof(*audio_ports));
+    for (i = 0; i < ports; i++) {
+        const jack_port_t *dport = JACK_jack_port_by_name(client, devports[i]);
+        const char *type = JACK_jack_port_type(dport);
+        const int len = SDL_strlen(type);
+        /* See if type ends with "audio" */
+        if (len >= 5 && !SDL_memcmp(type + len - 5, "audio", 5)) {
+            audio_ports[channels++] = i;
+        }
+    }
+    if (channels == 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("No physical JACK ports available");
+    }
+
+    /* !!! FIXME: docs say about buffer size: "This size may change, clients that depend on it must register a bufsize_callback so they will be notified if it does." */
+
+    /* Jack pretty much demands what it wants. */
+    this->spec.format = AUDIO_F32SYS;
+    this->spec.freq = JACK_jack_get_sample_rate(client);
+    this->spec.channels = channels;
+    this->spec.samples = JACK_jack_get_buffer_size(client);
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    this->hidden->iosem = SDL_CreateSemaphore(0);
+    if (!this->hidden->iosem) {
+        SDL_free(audio_ports);
+        return -1; /* error was set by SDL_CreateSemaphore */
+    }
+
+    this->hidden->iobuffer = (float *)SDL_calloc(1, this->spec.size);
+    if (!this->hidden->iobuffer) {
+        SDL_free(audio_ports);
+        return SDL_OutOfMemory();
+    }
+
+    /* Build SDL's ports, which we will connect to the device ports. */
+    this->hidden->sdlports = (jack_port_t **)SDL_calloc(channels, sizeof(jack_port_t *));
+    if (this->hidden->sdlports == NULL) {
+        SDL_free(audio_ports);
+        return SDL_OutOfMemory();
+    }
+
+    for (i = 0; i < channels; i++) {
+        char portname[32];
+        (void)SDL_snprintf(portname, sizeof(portname), "sdl_jack_%s_%d", sdlportstr, i);
+        this->hidden->sdlports[i] = JACK_jack_port_register(client, portname, JACK_DEFAULT_AUDIO_TYPE, sdlportflags, 0);
+        if (this->hidden->sdlports[i] == NULL) {
+            SDL_free(audio_ports);
+            return SDL_SetError("jack_port_register failed");
+        }
+    }
+
+    if (JACK_jack_set_process_callback(client, callback, this) != 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("JACK: Couldn't set process callback");
+    }
+
+    JACK_jack_on_shutdown(client, jackShutdownCallback, this);
+
+    if (JACK_jack_activate(client) != 0) {
+        SDL_free(audio_ports);
+        return SDL_SetError("Failed to activate JACK client");
+    }
+
+    /* once activated, we can connect all the ports. */
+    for (i = 0; i < channels; i++) {
+        const char *sdlport = JACK_jack_port_name(this->hidden->sdlports[i]);
+        const char *srcport = iscapture ? devports[audio_ports[i]] : sdlport;
+        const char *dstport = iscapture ? sdlport : devports[audio_ports[i]];
+        if (JACK_jack_connect(client, srcport, dstport) != 0) {
+            SDL_free(audio_ports);
+            return SDL_SetError("Couldn't connect JACK ports: %s => %s", srcport, dstport);
+        }
+    }
+
+    /* don't need these anymore. */
+    JACK_jack_free(devports);
+    SDL_free(audio_ports);
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static void JACK_Deinitialize(void)
+{
+    UnloadJackLibrary();
+}
+
+static SDL_bool JACK_Init(SDL_AudioDriverImpl *impl)
+{
+    if (LoadJackLibrary() < 0) {
+        return SDL_FALSE;
+    } else {
+        /* Make sure a JACK server is running and available. */
+        jack_status_t status;
+        jack_client_t *client = JACK_jack_client_open("SDL", JackNoStartServer, &status, NULL);
+        if (client == NULL) {
+            UnloadJackLibrary();
+            return SDL_FALSE;
+        }
+        JACK_jack_client_close(client);
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = JACK_OpenDevice;
+    impl->WaitDevice = JACK_WaitDevice;
+    impl->GetDeviceBuf = JACK_GetDeviceBuf;
+    impl->CloseDevice = JACK_CloseDevice;
+    impl->Deinitialize = JACK_Deinitialize;
+    impl->CaptureFromDevice = JACK_CaptureFromDevice;
+    impl->FlushCapture = JACK_FlushCapture;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap JACK_bootstrap = {
+    "jack", "JACK Audio Connection Kit", JACK_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_JACK */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/jack/SDL_jackaudio.h

@@ -0,0 +1,41 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#ifndef SDL_jackaudio_h_
+#define SDL_jackaudio_h_
+
+#include <jack/jack.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    jack_client_t *client;
+    SDL_sem *iosem;
+    float *iobuffer;
+    jack_port_t **sdlports;
+};
+
+#endif /* SDL_jackaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/n3ds/SDL_n3dsaudio.c

@@ -0,0 +1,347 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_N3DS
+
+#include "SDL_audio.h"
+
+/* N3DS Audio driver */
+
+#include "../SDL_sysaudio.h"
+#include "SDL_n3dsaudio.h"
+#include "SDL_timer.h"
+
+#define N3DSAUDIO_DRIVER_NAME "n3ds"
+
+static dspHookCookie dsp_hook;
+static SDL_AudioDevice *audio_device;
+
+static void FreePrivateData(_THIS);
+static int FindAudioFormat(_THIS);
+
+static SDL_INLINE void contextLock(_THIS)
+{
+    LightLock_Lock(&this->hidden->lock);
+}
+
+static SDL_INLINE void contextUnlock(_THIS)
+{
+    LightLock_Unlock(&this->hidden->lock);
+}
+
+static void N3DSAUD_LockAudio(_THIS)
+{
+    contextLock(this);
+}
+
+static void N3DSAUD_UnlockAudio(_THIS)
+{
+    contextUnlock(this);
+}
+
+static void N3DSAUD_DspHook(DSP_HookType hook)
+{
+    if (hook == DSPHOOK_ONCANCEL) {
+        contextLock(audio_device);
+        audio_device->hidden->isCancelled = SDL_TRUE;
+        SDL_AtomicSet(&audio_device->enabled, SDL_FALSE);
+        CondVar_Broadcast(&audio_device->hidden->cv);
+        contextUnlock(audio_device);
+    }
+}
+
+static void AudioFrameFinished(void *device)
+{
+    bool shouldBroadcast = false;
+    unsigned i;
+    SDL_AudioDevice *this = (SDL_AudioDevice *)device;
+
+    contextLock(this);
+
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        if (this->hidden->waveBuf[i].status == NDSP_WBUF_DONE) {
+            this->hidden->waveBuf[i].status = NDSP_WBUF_FREE;
+            shouldBroadcast = SDL_TRUE;
+        }
+    }
+
+    if (shouldBroadcast) {
+        CondVar_Broadcast(&this->hidden->cv);
+    }
+
+    contextUnlock(this);
+}
+
+static int N3DSAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    Result ndsp_init_res;
+    Uint8 *data_vaddr;
+    float mix[12];
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*this->hidden));
+
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    /* Initialise the DSP service */
+    ndsp_init_res = ndspInit();
+    if (R_FAILED(ndsp_init_res)) {
+        if ((R_SUMMARY(ndsp_init_res) == RS_NOTFOUND) && (R_MODULE(ndsp_init_res) == RM_DSP)) {
+            SDL_SetError("DSP init failed: dspfirm.cdc missing!");
+        } else {
+            SDL_SetError("DSP init failed. Error code: 0x%lX", ndsp_init_res);
+        }
+        return -1;
+    }
+
+    /* Initialise internal state */
+    LightLock_Init(&this->hidden->lock);
+    CondVar_Init(&this->hidden->cv);
+
+    if (this->spec.channels > 2) {
+        this->spec.channels = 2;
+    }
+
+    /* Should not happen but better be safe. */
+    if (FindAudioFormat(this) < 0) {
+        return SDL_SetError("No supported audio format found.");
+    }
+
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    if (this->spec.size >= SDL_MAX_UINT32 / 2) {
+        return SDL_SetError("Mixing buffer is too large.");
+    }
+
+    this->hidden->mixlen = this->spec.size;
+    this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->spec.size);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+
+    data_vaddr = (Uint8 *)linearAlloc(this->hidden->mixlen * NUM_BUFFERS);
+    if (data_vaddr == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    SDL_memset(data_vaddr, 0, this->hidden->mixlen * NUM_BUFFERS);
+    DSP_FlushDataCache(data_vaddr, this->hidden->mixlen * NUM_BUFFERS);
+
+    this->hidden->nextbuf = 0;
+    this->hidden->channels = this->spec.channels;
+    this->hidden->samplerate = this->spec.freq;
+
+    ndspChnReset(0);
+
+    ndspChnSetInterp(0, NDSP_INTERP_LINEAR);
+    ndspChnSetRate(0, this->spec.freq);
+    ndspChnSetFormat(0, this->hidden->format);
+
+    SDL_memset(mix, 0, sizeof(mix));
+    mix[0] = 1.0;
+    mix[1] = 1.0;
+    ndspChnSetMix(0, mix);
+
+    SDL_memset(this->hidden->waveBuf, 0, sizeof(ndspWaveBuf) * NUM_BUFFERS);
+
+    for (unsigned i = 0; i < NUM_BUFFERS; i++) {
+        this->hidden->waveBuf[i].data_vaddr = data_vaddr;
+        this->hidden->waveBuf[i].nsamples = this->hidden->mixlen / this->hidden->bytePerSample;
+        data_vaddr += this->hidden->mixlen;
+    }
+
+    /* Setup callback */
+    audio_device = this;
+    ndspSetCallback(AudioFrameFinished, this);
+    dspHook(&dsp_hook, N3DSAUD_DspHook);
+
+    return 0;
+}
+
+static int N3DSAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    /* Delay to make this sort of simulate real audio input. */
+    SDL_Delay((this->spec.samples * 1000) / this->spec.freq);
+
+    /* always return a full buffer of silence. */
+    SDL_memset(buffer, this->spec.silence, buflen);
+    return buflen;
+}
+
+static void N3DSAUDIO_PlayDevice(_THIS)
+{
+    size_t nextbuf;
+    size_t sampleLen;
+    contextLock(this);
+
+    nextbuf = this->hidden->nextbuf;
+    sampleLen = this->hidden->mixlen;
+
+    if (this->hidden->isCancelled ||
+        this->hidden->waveBuf[nextbuf].status != NDSP_WBUF_FREE) {
+        contextUnlock(this);
+        return;
+    }
+
+    this->hidden->nextbuf = (nextbuf + 1) % NUM_BUFFERS;
+
+    contextUnlock(this);
+
+    memcpy((void *)this->hidden->waveBuf[nextbuf].data_vaddr,
+           this->hidden->mixbuf, sampleLen);
+    DSP_FlushDataCache(this->hidden->waveBuf[nextbuf].data_vaddr, sampleLen);
+
+    ndspChnWaveBufAdd(0, &this->hidden->waveBuf[nextbuf]);
+}
+
+static void N3DSAUDIO_WaitDevice(_THIS)
+{
+    contextLock(this);
+    while (!this->hidden->isCancelled &&
+           this->hidden->waveBuf[this->hidden->nextbuf].status != NDSP_WBUF_FREE) {
+        CondVar_Wait(&this->hidden->cv, &this->hidden->lock);
+    }
+    contextUnlock(this);
+}
+
+static Uint8 *N3DSAUDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static void N3DSAUDIO_CloseDevice(_THIS)
+{
+    contextLock(this);
+
+    dspUnhook(&dsp_hook);
+    ndspSetCallback(NULL, NULL);
+
+    if (!this->hidden->isCancelled) {
+        ndspChnReset(0);
+        memset(this->hidden->waveBuf, 0, sizeof(ndspWaveBuf) * NUM_BUFFERS);
+        CondVar_Broadcast(&this->hidden->cv);
+    }
+
+    contextUnlock(this);
+
+    ndspExit();
+
+    FreePrivateData(this);
+}
+
+static void N3DSAUDIO_ThreadInit(_THIS)
+{
+    s32 current_priority;
+    svcGetThreadPriority(&current_priority, CUR_THREAD_HANDLE);
+    current_priority--;
+    /* 0x18 is reserved for video, 0x30 is the default for main thread */
+    current_priority = SDL_clamp(current_priority, 0x19, 0x2F);
+    svcSetThreadPriority(CUR_THREAD_HANDLE, current_priority);
+}
+
+static SDL_bool N3DSAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->OpenDevice = N3DSAUDIO_OpenDevice;
+    impl->PlayDevice = N3DSAUDIO_PlayDevice;
+    impl->WaitDevice = N3DSAUDIO_WaitDevice;
+    impl->GetDeviceBuf = N3DSAUDIO_GetDeviceBuf;
+    impl->CloseDevice = N3DSAUDIO_CloseDevice;
+    impl->ThreadInit = N3DSAUDIO_ThreadInit;
+    impl->LockDevice = N3DSAUD_LockAudio;
+    impl->UnlockDevice = N3DSAUD_UnlockAudio;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+
+    /* Should be possible, but micInit would fail */
+    impl->HasCaptureSupport = SDL_FALSE;
+    impl->CaptureFromDevice = N3DSAUDIO_CaptureFromDevice;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap N3DSAUDIO_bootstrap = {
+    N3DSAUDIO_DRIVER_NAME,
+    "SDL N3DS audio driver",
+    N3DSAUDIO_Init,
+    0
+};
+
+/**
+ * Cleans up all allocated memory, safe to call with null pointers
+ */
+static void FreePrivateData(_THIS)
+{
+    if (!this->hidden) {
+        return;
+    }
+
+    if (this->hidden->waveBuf[0].data_vaddr) {
+        linearFree((void *)this->hidden->waveBuf[0].data_vaddr);
+    }
+
+    if (this->hidden->mixbuf) {
+        SDL_free(this->hidden->mixbuf);
+        this->hidden->mixbuf = NULL;
+    }
+
+    SDL_free(this->hidden);
+    this->hidden = NULL;
+}
+
+static int FindAudioFormat(_THIS)
+{
+    SDL_bool found_valid_format = SDL_FALSE;
+    Uint16 test_format = SDL_FirstAudioFormat(this->spec.format);
+
+    while (!found_valid_format && test_format) {
+        this->spec.format = test_format;
+        switch (test_format) {
+        case AUDIO_S8:
+            /* Signed 8-bit audio supported */
+            this->hidden->format = (this->spec.channels == 2) ? NDSP_FORMAT_STEREO_PCM8 : NDSP_FORMAT_MONO_PCM8;
+            this->hidden->isSigned = 1;
+            this->hidden->bytePerSample = this->spec.channels;
+            found_valid_format = SDL_TRUE;
+            break;
+        case AUDIO_S16:
+            /* Signed 16-bit audio supported */
+            this->hidden->format = (this->spec.channels == 2) ? NDSP_FORMAT_STEREO_PCM16 : NDSP_FORMAT_MONO_PCM16;
+            this->hidden->isSigned = 1;
+            this->hidden->bytePerSample = this->spec.channels * 2;
+            found_valid_format = SDL_TRUE;
+            break;
+        default:
+            test_format = SDL_NextAudioFormat();
+            break;
+        }
+    }
+
+    return found_valid_format ? 0 : -1;
+}
+
+#endif /* SDL_AUDIO_DRIVER_N3DS */
+
+/* vi: set sts=4 ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/n3ds/SDL_n3dsaudio.h

@@ -0,0 +1,50 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef _SDL_n3dsaudio_h_
+#define _SDL_n3dsaudio_h_
+
+#include <3ds.h>
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+#define NUM_BUFFERS 2 /* -- Don't lower this! */
+
+struct SDL_PrivateAudioData
+{
+    /* Speaker data */
+    Uint8 *mixbuf;
+    Uint32 mixlen;
+    Uint32 format;
+    Uint32 samplerate;
+    Uint32 channels;
+    Uint8 bytePerSample;
+    Uint32 isSigned;
+    Uint32 nextbuf;
+    ndspWaveBuf waveBuf[NUM_BUFFERS];
+    LightLock lock;
+    CondVar cv;
+    SDL_bool isCancelled;
+};
+
+#endif /* _SDL_n3dsaudio_h_ */
+/* vi: set sts=4 ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/nacl/SDL_naclaudio.c

@@ -0,0 +1,161 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_NACL
+
+#include "SDL_naclaudio.h"
+
+#include "SDL_audio.h"
+#include "SDL_mutex.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+
+#include "ppapi/c/pp_errors.h"
+#include "ppapi/c/pp_instance.h"
+#include "ppapi_simple/ps.h"
+#include "ppapi_simple/ps_interface.h"
+#include "ppapi_simple/ps_event.h"
+
+/* The tag name used by NACL audio */
+#define NACLAUDIO_DRIVER_NAME         "nacl"
+
+#define SAMPLE_FRAME_COUNT 4096
+
+/* Audio driver functions */
+static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data);
+
+/* FIXME: Make use of latency if needed */
+static void nacl_audio_callback(void* stream, uint32_t buffer_size, PP_TimeDelta latency, void* data)
+{
+    const int len = (int) buffer_size;
+    SDL_AudioDevice* _this = (SDL_AudioDevice*) data;
+    SDL_AudioCallback callback = _this->callbackspec.callback;
+
+    SDL_LockMutex(_this->mixer_lock);
+
+    /* Only do something if audio is enabled */
+    if (!SDL_AtomicGet(&_this->enabled) || SDL_AtomicGet(&_this->paused)) {
+        if (_this->stream) {
+            SDL_AudioStreamClear(_this->stream);
+        }
+        SDL_memset(stream, _this->spec.silence, len);
+    } else {
+        SDL_assert(_this->spec.size == len);
+
+        if (_this->stream == NULL) {  /* no conversion necessary. */
+            callback(_this->callbackspec.userdata, stream, len);
+        } else {  /* streaming/converting */
+            const int stream_len = _this->callbackspec.size;
+            while (SDL_AudioStreamAvailable(_this->stream) < len) {
+                callback(_this->callbackspec.userdata, _this->work_buffer, stream_len);
+                if (SDL_AudioStreamPut(_this->stream, _this->work_buffer, stream_len) == -1) {
+                    SDL_AudioStreamClear(_this->stream);
+                    SDL_AtomicSet(&_this->enabled, 0);
+                    break;
+                }
+            }
+
+            const int got = SDL_AudioStreamGet(_this->stream, stream, len);
+            SDL_assert((got < 0) || (got == len));
+            if (got != len) {
+                SDL_memset(stream, _this->spec.silence, len);
+            }
+        }
+    }
+
+    SDL_UnlockMutex(_this->mixer_lock);
+}
+
+static void NACLAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    const PPB_Core *core = PSInterfaceCore();
+    const PPB_Audio *ppb_audio = PSInterfaceAudio();
+    SDL_PrivateAudioData *hidden = (SDL_PrivateAudioData *) device->hidden;
+
+    ppb_audio->StopPlayback(hidden->audio);
+    core->ReleaseResource(hidden->audio);
+}
+
+static int NACLAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    PP_Instance instance = PSGetInstanceId();
+    const PPB_Audio *ppb_audio = PSInterfaceAudio();
+    const PPB_AudioConfig *ppb_audiocfg = PSInterfaceAudioConfig();
+
+    private = (SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*private));
+    if (private == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    _this->spec.freq = 44100;
+    _this->spec.format = AUDIO_S16LSB;
+    _this->spec.channels = 2;
+    _this->spec.samples = ppb_audiocfg->RecommendSampleFrameCount(
+        instance,
+        PP_AUDIOSAMPLERATE_44100,
+        SAMPLE_FRAME_COUNT);
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&_this->spec);
+
+    private->audio = ppb_audio->Create(
+        instance,
+        ppb_audiocfg->CreateStereo16Bit(instance, PP_AUDIOSAMPLERATE_44100, _this->spec.samples),
+        nacl_audio_callback,
+        _this);
+
+    /* Start audio playback while we are still on the main thread. */
+    ppb_audio->StartPlayback(private->audio);
+
+    return 0;
+}
+
+static SDL_bool NACLAUDIO_Init(SDL_AudioDriverImpl * impl)
+{
+    if (PSGetInstanceId() == 0) {
+        return SDL_FALSE;
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = NACLAUDIO_OpenDevice;
+    impl->CloseDevice = NACLAUDIO_CloseDevice;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->ProvidesOwnCallbackThread = SDL_TRUE;
+    /*
+     *    impl->WaitDevice = NACLAUDIO_WaitDevice;
+     *    impl->GetDeviceBuf = NACLAUDIO_GetDeviceBuf;
+     *    impl->PlayDevice = NACLAUDIO_PlayDevice;
+     *    impl->Deinitialize = NACLAUDIO_Deinitialize;
+     */
+
+    return SDL_TRUE;
+}
+
+AudioBootStrap NACLAUDIO_bootstrap = {
+    NACLAUDIO_DRIVER_NAME, "SDL NaCl Audio Driver",
+    NACLAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_NACL */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/nacl/SDL_naclaudio.h

@@ -0,0 +1,42 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#ifndef SDL_naclaudio_h_
+#define SDL_naclaudio_h_
+
+#include "SDL_audio.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_mutex.h"
+
+#include "ppapi/c/ppb_audio.h"
+
+#define _THIS  SDL_AudioDevice *_this
+#define private _this->hidden
+
+typedef struct SDL_PrivateAudioData {
+    PP_Resource audio;
+} SDL_PrivateAudioData;
+
+#endif /* SDL_naclaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/nas/SDL_nasaudio.c

@@ -0,0 +1,443 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_NAS
+
+/* Allow access to a raw mixing buffer */
+
+#include <signal.h>
+#include <unistd.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "SDL_loadso.h"
+#include "../SDL_audio_c.h"
+#include "SDL_nasaudio.h"
+
+static void (*NAS_AuCloseServer) (AuServer *);
+static void (*NAS_AuNextEvent) (AuServer *, AuBool, AuEvent *);
+static AuBool(*NAS_AuDispatchEvent) (AuServer *, AuEvent *);
+static void (*NAS_AuHandleEvents) (AuServer *);
+static AuFlowID(*NAS_AuCreateFlow) (AuServer *, AuStatus *);
+static void (*NAS_AuStartFlow) (AuServer *, AuFlowID, AuStatus *);
+static void (*NAS_AuSetElements)
+  (AuServer *, AuFlowID, AuBool, int, AuElement *, AuStatus *);
+static void (*NAS_AuWriteElement)
+  (AuServer *, AuFlowID, int, AuUint32, AuPointer, AuBool, AuStatus *);
+static AuUint32 (*NAS_AuReadElement)
+  (AuServer *, AuFlowID, int, AuUint32, AuPointer, AuStatus *);
+static AuServer *(*NAS_AuOpenServer)
+  (_AuConst char *, int, _AuConst char *, int, _AuConst char *, char **);
+static AuEventHandlerRec *(*NAS_AuRegisterEventHandler)
+  (AuServer *, AuMask, int, AuID, AuEventHandlerCallback, AuPointer);
+
+
+#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
+
+static const char *nas_library = SDL_AUDIO_DRIVER_NAS_DYNAMIC;
+static void *nas_handle = NULL;
+
+static int load_nas_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(nas_handle, fn);
+    if (*addr == NULL) {
+        return 0;
+    }
+    return 1;
+}
+
+/* cast funcs to char* first, to please GCC's strict aliasing rules. */
+#define SDL_NAS_SYM(x) \
+    if (!load_nas_sym(#x, (void **) (char *) &NAS_##x)) return -1
+#else
+#define SDL_NAS_SYM(x) NAS_##x = x
+#endif
+
+static int load_nas_syms(void)
+{
+    SDL_NAS_SYM(AuCloseServer);
+    SDL_NAS_SYM(AuNextEvent);
+    SDL_NAS_SYM(AuDispatchEvent);
+    SDL_NAS_SYM(AuHandleEvents);
+    SDL_NAS_SYM(AuCreateFlow);
+    SDL_NAS_SYM(AuStartFlow);
+    SDL_NAS_SYM(AuSetElements);
+    SDL_NAS_SYM(AuWriteElement);
+    SDL_NAS_SYM(AuReadElement);
+    SDL_NAS_SYM(AuOpenServer);
+    SDL_NAS_SYM(AuRegisterEventHandler);
+    return 0;
+}
+
+#undef SDL_NAS_SYM
+
+#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
+
+static void UnloadNASLibrary(void)
+{
+    if (nas_handle != NULL) {
+        SDL_UnloadObject(nas_handle);
+        nas_handle = NULL;
+    }
+}
+
+static int LoadNASLibrary(void)
+{
+    int retval = 0;
+    if (nas_handle == NULL) {
+        nas_handle = SDL_LoadObject(nas_library);
+        if (nas_handle == NULL) {
+            /* Copy error string so we can use it in a new SDL_SetError(). */
+            const char *origerr = SDL_GetError();
+            const size_t len = SDL_strlen(origerr) + 1;
+            char *err = SDL_stack_alloc(char, len);
+            SDL_strlcpy(err, origerr, len);
+            SDL_SetError("NAS: SDL_LoadObject('%s') failed: %s", nas_library, err);
+            SDL_stack_free(err);
+            retval = -1;
+        } else {
+            retval = load_nas_syms();
+            if (retval < 0) {
+                UnloadNASLibrary();
+            }
+        }
+    }
+    return retval;
+}
+
+#else
+
+static void UnloadNASLibrary(void)
+{
+}
+
+static int LoadNASLibrary(void)
+{
+    load_nas_syms();
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_NAS_DYNAMIC */
+
+/* This function waits until it is possible to write a full sound buffer */
+static void NAS_WaitDevice(_THIS)
+{
+    while (this->hidden->buf_free < this->hidden->mixlen) {
+        AuEvent ev;
+        NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev);
+        NAS_AuDispatchEvent(this->hidden->aud, &ev);
+    }
+}
+
+static void NAS_PlayDevice(_THIS)
+{
+    while (this->hidden->mixlen > this->hidden->buf_free) {
+        /*
+         * We think the buffer is full? Yikes! Ask the server for events,
+         *  in the hope that some of them is LowWater events telling us more
+         *  of the buffer is free now than what we think.
+         */
+        AuEvent ev;
+        NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev);
+        NAS_AuDispatchEvent(this->hidden->aud, &ev);
+    }
+    this->hidden->buf_free -= this->hidden->mixlen;
+
+    /* Write the audio data */
+    NAS_AuWriteElement(this->hidden->aud, this->hidden->flow, 0,
+                       this->hidden->mixlen, this->hidden->mixbuf, AuFalse,
+                       NULL);
+
+    this->hidden->written += this->hidden->mixlen;
+
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen);
+#endif
+}
+
+static Uint8 *NAS_GetDeviceBuf(_THIS)
+{
+    return (this->hidden->mixbuf);
+}
+
+static int NAS_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    int retval;
+
+    while (SDL_TRUE) {
+        /* just keep the event queue moving and the server chattering. */
+        NAS_AuHandleEvents(h->aud);
+
+        retval = (int) NAS_AuReadElement(h->aud, h->flow, 1, buflen, buffer, NULL);
+        /*printf("read %d capture bytes\n", (int) retval);*/
+        if (retval == 0) {
+            SDL_Delay(10);  /* don't burn the CPU if we're waiting for data. */
+        } else {
+            break;
+        }
+    }
+
+    return retval;
+}
+
+static void NAS_FlushCapture(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    AuUint32 total = 0;
+    AuUint32 br;
+    Uint8 buf[512];
+
+    do {
+        /* just keep the event queue moving and the server chattering. */
+        NAS_AuHandleEvents(h->aud);
+        br = NAS_AuReadElement(h->aud, h->flow, 1, sizeof(buf), buf, NULL);
+        /*printf("flushed %d capture bytes\n", (int) br);*/
+        total += br;
+    } while ((br == sizeof(buf)) && (total < this->spec.size));
+}
+
+static void NAS_CloseDevice(_THIS)
+{
+    if (this->hidden->aud) {
+        NAS_AuCloseServer(this->hidden->aud);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static AuBool event_handler(AuServer * aud, AuEvent * ev, AuEventHandlerRec * hnd)
+{
+    SDL_AudioDevice *this = (SDL_AudioDevice *) hnd->data;
+    struct SDL_PrivateAudioData *h = this->hidden;
+    if (this->iscapture) {
+        return AuTrue;  /* we don't (currently) care about any of this for capture devices */
+    }
+
+    switch (ev->type) {
+    case AuEventTypeElementNotify:
+        {
+            AuElementNotifyEvent *event = (AuElementNotifyEvent *) ev;
+
+            switch (event->kind) {
+            case AuElementNotifyKindLowWater:
+                if (h->buf_free >= 0) {
+                    h->really += event->num_bytes;
+                    gettimeofday(&h->last_tv, 0);
+                    h->buf_free += event->num_bytes;
+                } else {
+                    h->buf_free = event->num_bytes;
+                }
+                break;
+            case AuElementNotifyKindState:
+                switch (event->cur_state) {
+                case AuStatePause:
+                    if (event->reason != AuReasonUser) {
+                        if (h->buf_free >= 0) {
+                            h->really += event->num_bytes;
+                            gettimeofday(&h->last_tv, 0);
+                            h->buf_free += event->num_bytes;
+                        } else {
+                            h->buf_free = event->num_bytes;
+                        }
+                    }
+                    break;
+                }
+            }
+        }
+    }
+    return AuTrue;
+}
+
+static AuDeviceID find_device(_THIS)
+{
+    /* These "Au" things are all macros, not functions... */
+    struct SDL_PrivateAudioData *h = this->hidden;
+    const unsigned int devicekind = this->iscapture ? AuComponentKindPhysicalInput : AuComponentKindPhysicalOutput;
+    const int numdevs = AuServerNumDevices(h->aud);
+    const int nch = this->spec.channels;
+    int i;
+
+    /* Try to find exact match on channels first... */
+    for (i = 0; i < numdevs; i++) {
+        const AuDeviceAttributes *dev = AuServerDevice(h->aud, i);
+        if ((AuDeviceKind(dev) == devicekind) && (AuDeviceNumTracks(dev) == nch)) {
+            return AuDeviceIdentifier(dev);
+        }
+    }
+
+    /* Take anything, then... */
+    for (i = 0; i < numdevs; i++) {
+        const AuDeviceAttributes *dev = AuServerDevice(h->aud, i);
+        if (AuDeviceKind(dev) == devicekind) {
+            this->spec.channels = AuDeviceNumTracks(dev);
+            return AuDeviceIdentifier(dev);
+        }
+    }
+    return AuNone;
+}
+
+static int NAS_OpenDevice(_THIS, const char *devname)
+{
+    AuElement elms[3];
+    int buffer_size;
+    SDL_bool iscapture = this->iscapture;
+    SDL_AudioFormat test_format, format = 0;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_U8:
+            format = AuFormatLinearUnsigned8;
+            break;
+        case AUDIO_S8:
+            format = AuFormatLinearSigned8;
+            break;
+        case AUDIO_U16LSB:
+            format = AuFormatLinearUnsigned16LSB;
+            break;
+        case AUDIO_U16MSB:
+            format = AuFormatLinearUnsigned16MSB;
+            break;
+        case AUDIO_S16LSB:
+            format = AuFormatLinearSigned16LSB;
+            break;
+        case AUDIO_S16MSB:
+            format = AuFormatLinearSigned16MSB;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "nas");
+    }
+    this->spec.format = test_format;
+
+    this->hidden->aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
+    if (this->hidden->aud == 0) {
+        return SDL_SetError("NAS: Couldn't open connection to NAS server");
+    }
+
+    this->hidden->dev = find_device(this);
+    if ((this->hidden->dev == AuNone)
+        || (!(this->hidden->flow = NAS_AuCreateFlow(this->hidden->aud, 0)))) {
+        return SDL_SetError("NAS: Couldn't find a fitting device on NAS server");
+    }
+
+    buffer_size = this->spec.freq;
+    if (buffer_size < 4096)
+        buffer_size = 4096;
+
+    if (buffer_size > 32768)
+        buffer_size = 32768;    /* So that the buffer won't get unmanageably big. */
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    if (iscapture) {
+        AuMakeElementImportDevice(elms, this->spec.freq, this->hidden->dev,
+                                  AuUnlimitedSamples, 0, NULL);
+        AuMakeElementExportClient(elms + 1, 0, this->spec.freq, format,
+                                  this->spec.channels, AuTrue, buffer_size,
+                                  buffer_size, 0, NULL);
+    } else {
+        AuMakeElementImportClient(elms, this->spec.freq, format,
+                                  this->spec.channels, AuTrue, buffer_size,
+                                  buffer_size / 4, 0, NULL);
+        AuMakeElementExportDevice(elms + 1, 0, this->hidden->dev, this->spec.freq,
+                                  AuUnlimitedSamples, 0, NULL);
+    }
+
+    NAS_AuSetElements(this->hidden->aud, this->hidden->flow, AuTrue,
+                      2, elms, NULL);
+
+    NAS_AuRegisterEventHandler(this->hidden->aud, AuEventHandlerIDMask, 0,
+                               this->hidden->flow, event_handler,
+                               (AuPointer) this);
+
+    NAS_AuStartFlow(this->hidden->aud, this->hidden->flow, NULL);
+
+    /* Allocate mixing buffer */
+    if (!iscapture) {
+        this->hidden->mixlen = this->spec.size;
+        this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
+        if (this->hidden->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+    }
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static void NAS_Deinitialize(void)
+{
+    UnloadNASLibrary();
+}
+
+static SDL_bool NAS_Init(SDL_AudioDriverImpl * impl)
+{
+    if (LoadNASLibrary() < 0) {
+        return SDL_FALSE;
+    } else {
+        AuServer *aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
+        if (aud == NULL) {
+            SDL_SetError("NAS: AuOpenServer() failed (no audio server?)");
+            return SDL_FALSE;
+        }
+        NAS_AuCloseServer(aud);
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = NAS_OpenDevice;
+    impl->PlayDevice = NAS_PlayDevice;
+    impl->WaitDevice = NAS_WaitDevice;
+    impl->GetDeviceBuf = NAS_GetDeviceBuf;
+    impl->CaptureFromDevice = NAS_CaptureFromDevice;
+    impl->FlushCapture = NAS_FlushCapture;
+    impl->CloseDevice = NAS_CloseDevice;
+    impl->Deinitialize = NAS_Deinitialize;
+
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+AudioBootStrap NAS_bootstrap = {
+    "nas", "Network Audio System", NAS_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_NAS */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/nas/SDL_nasaudio.h

@@ -0,0 +1,56 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_nasaudio_h_
+#define SDL_nasaudio_h_
+
+#ifdef __sgi
+#include <nas/audiolib.h>
+#else
+#include <audio/audiolib.h>
+#endif
+#include <sys/time.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    AuServer *aud;
+    AuFlowID flow;
+    AuDeviceID dev;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    int written;
+    int really;
+    int bps;
+    struct timeval last_tv;
+    int buf_free;
+};
+#endif /* SDL_nasaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/netbsd/SDL_netbsdaudio.c

@@ -0,0 +1,334 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_NETBSD
+
+/*
+ * Driver for native NetBSD audio(4).
+ * nia@NetBSD.org
+ */
+
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <sys/audioio.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../../core/unix/SDL_poll.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "SDL_netbsdaudio.h"
+
+/* #define DEBUG_AUDIO */
+
+static void NETBSDAUDIO_DetectDevices(void)
+{
+    SDL_EnumUnixAudioDevices(0, NULL);
+}
+
+static void NETBSDAUDIO_Status(_THIS)
+{
+#ifdef DEBUG_AUDIO
+    /* *INDENT-OFF* */ /* clang-format off */
+    audio_info_t info;
+    const struct audio_prinfo *prinfo;
+
+    if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
+        fprintf(stderr, "AUDIO_GETINFO failed.\n");
+        return;
+    }
+
+    prinfo = this->iscapture ? &info.record : &info.play;
+
+    fprintf(stderr, "\n"
+            "[%s info]\n"
+            "buffer size	:   %d bytes\n"
+            "sample rate	:   %i Hz\n"
+            "channels	:   %i\n"
+            "precision	:   %i-bit\n"
+            "encoding	:   0x%x\n"
+            "seek		:   %i\n"
+            "sample count	:   %i\n"
+            "EOF count	:   %i\n"
+            "paused		:   %s\n"
+            "error occured	:   %s\n"
+            "waiting		:   %s\n"
+            "active		:   %s\n"
+            "",
+            this->iscapture ? "record" : "play",
+            prinfo->buffer_size,
+            prinfo->sample_rate,
+            prinfo->channels,
+            prinfo->precision,
+            prinfo->encoding,
+            prinfo->seek,
+            prinfo->samples,
+            prinfo->eof,
+            prinfo->pause ? "yes" : "no",
+            prinfo->error ? "yes" : "no",
+            prinfo->waiting ? "yes" : "no",
+            prinfo->active ? "yes" : "no");
+
+    fprintf(stderr, "\n"
+            "[audio info]\n"
+            "monitor_gain	:   %i\n"
+            "hw block size	:   %d bytes\n"
+            "hi watermark	:   %i\n"
+            "lo watermark	:   %i\n"
+            "audio mode	:   %s\n"
+            "",
+            info.monitor_gain,
+            info.blocksize,
+            info.hiwat, info.lowat,
+            (info.mode == AUMODE_PLAY) ? "PLAY"
+            : (info.mode = AUMODE_RECORD) ? "RECORD"
+            : (info.mode == AUMODE_PLAY_ALL ? "PLAY_ALL" : "?"));
+
+    fprintf(stderr, "\n"
+            "[audio spec]\n"
+            "format		:   0x%x\n"
+            "size		:   %u\n"
+            "",
+            this->spec.format,
+            this->spec.size);
+    /* *INDENT-ON* */ /* clang-format on */
+
+#endif /* DEBUG_AUDIO */
+}
+
+static void NETBSDAUDIO_PlayDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    int written;
+
+    /* Write the audio data */
+    written = write(h->audio_fd, h->mixbuf, h->mixlen);
+    if (written == -1) {
+        /* Non recoverable error has occurred. It should be reported!!! */
+        SDL_OpenedAudioDeviceDisconnected(this);
+        perror("audio");
+        return;
+    }
+
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+}
+
+static Uint8 *NETBSDAUDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static int NETBSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
+{
+    Uint8 *buffer = (Uint8 *)_buffer;
+    int br;
+
+    br = read(this->hidden->audio_fd, buffer, buflen);
+    if (br == -1) {
+        /* Non recoverable error has occurred. It should be reported!!! */
+        perror("audio");
+        return -1;
+    }
+
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Captured %d bytes of audio data\n", br);
+#endif
+    return 0;
+}
+
+static void NETBSDAUDIO_FlushCapture(_THIS)
+{
+    audio_info_t info;
+    size_t remain;
+    Uint8 buf[512];
+
+    if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
+        return; /* oh well. */
+    }
+
+    remain = (size_t)(info.record.samples * (SDL_AUDIO_BITSIZE(this->spec.format) / 8));
+    while (remain > 0) {
+        const size_t len = SDL_min(sizeof(buf), remain);
+        const int br = read(this->hidden->audio_fd, buf, len);
+        if (br <= 0) {
+            return; /* oh well. */
+        }
+        remain -= br;
+    }
+}
+
+static void NETBSDAUDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->audio_fd >= 0) {
+        close(this->hidden->audio_fd);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int NETBSDAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    SDL_bool iscapture = this->iscapture;
+    SDL_AudioFormat test_format;
+    int encoding = AUDIO_ENCODING_NONE;
+    audio_info_t info, hwinfo;
+    struct audio_prinfo *prinfo = iscapture ? &info.record : &info.play;
+
+    /* We don't care what the devname is...we'll try to open anything. */
+    /*  ...but default to first name in the list... */
+    if (devname == NULL) {
+        devname = SDL_GetAudioDeviceName(0, iscapture);
+        if (devname == NULL) {
+            return SDL_SetError("No such audio device");
+        }
+    }
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *) SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Open the audio device */
+    this->hidden->audio_fd = open(devname, (iscapture ? O_RDONLY : O_WRONLY) | O_CLOEXEC);
+    if (this->hidden->audio_fd < 0) {
+        return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
+    }
+
+    AUDIO_INITINFO(&info);
+
+#ifdef AUDIO_GETFORMAT /* Introduced in NetBSD 9.0 */
+    if (ioctl(this->hidden->audio_fd, AUDIO_GETFORMAT, &hwinfo) != -1) {
+        /*
+         * Use the device's native sample rate so the kernel doesn't have to
+         * resample.
+         */
+        this->spec.freq = iscapture ? hwinfo.record.sample_rate : hwinfo.play.sample_rate;
+    }
+#endif
+
+    prinfo->sample_rate = this->spec.freq;
+    prinfo->channels = this->spec.channels;
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_U8:
+            encoding = AUDIO_ENCODING_ULINEAR;
+            break;
+        case AUDIO_S8:
+            encoding = AUDIO_ENCODING_SLINEAR;
+            break;
+        case AUDIO_S16LSB:
+            encoding = AUDIO_ENCODING_SLINEAR_LE;
+            break;
+        case AUDIO_S16MSB:
+            encoding = AUDIO_ENCODING_SLINEAR_BE;
+            break;
+        case AUDIO_U16LSB:
+            encoding = AUDIO_ENCODING_ULINEAR_LE;
+            break;
+        case AUDIO_U16MSB:
+            encoding = AUDIO_ENCODING_ULINEAR_BE;
+            break;
+        case AUDIO_S32LSB:
+            encoding = AUDIO_ENCODING_SLINEAR_LE;
+            break;
+        case AUDIO_S32MSB:
+            encoding = AUDIO_ENCODING_SLINEAR_BE;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "netbsd");
+    }
+    prinfo->encoding = encoding;
+    prinfo->precision = SDL_AUDIO_BITSIZE(test_format);
+
+    info.hiwat = 5;
+    info.lowat = 3;
+    if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) < 0) {
+        return SDL_SetError("AUDIO_SETINFO failed for %s: %s", devname, strerror(errno));
+    }
+
+    if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
+        return SDL_SetError("AUDIO_GETINFO failed for %s: %s", devname, strerror(errno));
+    }
+
+    /* Final spec used for the device. */
+    this->spec.format = test_format;
+    this->spec.freq = prinfo->sample_rate;
+    this->spec.channels = prinfo->channels;
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    if (!iscapture) {
+        /* Allocate mixing buffer */
+        this->hidden->mixlen = this->spec.size;
+        this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
+        if (this->hidden->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+    }
+
+    NETBSDAUDIO_Status(this);
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static SDL_bool NETBSDAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->DetectDevices = NETBSDAUDIO_DetectDevices;
+    impl->OpenDevice = NETBSDAUDIO_OpenDevice;
+    impl->PlayDevice = NETBSDAUDIO_PlayDevice;
+    impl->GetDeviceBuf = NETBSDAUDIO_GetDeviceBuf;
+    impl->CloseDevice = NETBSDAUDIO_CloseDevice;
+    impl->CaptureFromDevice = NETBSDAUDIO_CaptureFromDevice;
+    impl->FlushCapture = NETBSDAUDIO_FlushCapture;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap NETBSDAUDIO_bootstrap = {
+    "netbsd", "NetBSD audio", NETBSDAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_NETBSD */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/netbsd/SDL_netbsdaudio.h

@@ -0,0 +1,48 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_netbsdaudio_h_
+#define SDL_netbsdaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    int audio_fd;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    /* Support for audio timing using a timer, in addition to SDL_IOReady() */
+    float frame_ticks;
+    float next_frame;
+};
+
+#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
+
+#endif /* SDL_netbsdaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/openslES/SDL_openslES.c

@@ -0,0 +1,775 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_OPENSLES
+
+/* For more discussion of low latency audio on Android, see this:
+   https://googlesamples.github.io/android-audio-high-performance/guides/opensl_es.html
+*/
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "../../core/android/SDL_android.h"
+#include "SDL_openslES.h"
+
+/* for native audio */
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+
+#include <android/log.h>
+
+#if 0
+#define LOG_TAG   "SDL_openslES"
+#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
+#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
+//#define LOGV(...)  __android_log_print(ANDROID_LOG_VERBOSE,LOG_TAG,__VA_ARGS__)
+#define LOGV(...)
+#else
+#define LOGE(...)
+#define LOGI(...)
+#define LOGV(...)
+#endif
+
+/*
+#define SL_SPEAKER_FRONT_LEFT            ((SLuint32) 0x00000001)
+#define SL_SPEAKER_FRONT_RIGHT           ((SLuint32) 0x00000002)
+#define SL_SPEAKER_FRONT_CENTER          ((SLuint32) 0x00000004)
+#define SL_SPEAKER_LOW_FREQUENCY         ((SLuint32) 0x00000008)
+#define SL_SPEAKER_BACK_LEFT             ((SLuint32) 0x00000010)
+#define SL_SPEAKER_BACK_RIGHT            ((SLuint32) 0x00000020)
+#define SL_SPEAKER_FRONT_LEFT_OF_CENTER  ((SLuint32) 0x00000040)
+#define SL_SPEAKER_FRONT_RIGHT_OF_CENTER ((SLuint32) 0x00000080)
+#define SL_SPEAKER_BACK_CENTER           ((SLuint32) 0x00000100)
+#define SL_SPEAKER_SIDE_LEFT             ((SLuint32) 0x00000200)
+#define SL_SPEAKER_SIDE_RIGHT            ((SLuint32) 0x00000400)
+#define SL_SPEAKER_TOP_CENTER            ((SLuint32) 0x00000800)
+#define SL_SPEAKER_TOP_FRONT_LEFT        ((SLuint32) 0x00001000)
+#define SL_SPEAKER_TOP_FRONT_CENTER      ((SLuint32) 0x00002000)
+#define SL_SPEAKER_TOP_FRONT_RIGHT       ((SLuint32) 0x00004000)
+#define SL_SPEAKER_TOP_BACK_LEFT         ((SLuint32) 0x00008000)
+#define SL_SPEAKER_TOP_BACK_CENTER       ((SLuint32) 0x00010000)
+#define SL_SPEAKER_TOP_BACK_RIGHT        ((SLuint32) 0x00020000)
+*/
+#define SL_ANDROID_SPEAKER_STEREO (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT)
+#define SL_ANDROID_SPEAKER_QUAD   (SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT)
+#define SL_ANDROID_SPEAKER_5DOT1  (SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY)
+#define SL_ANDROID_SPEAKER_7DOT1  (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT | SL_SPEAKER_SIDE_RIGHT)
+
+/* engine interfaces */
+static SLObjectItf engineObject = NULL;
+static SLEngineItf engineEngine = NULL;
+
+/* output mix interfaces */
+static SLObjectItf outputMixObject = NULL;
+
+/* buffer queue player interfaces */
+static SLObjectItf bqPlayerObject = NULL;
+static SLPlayItf bqPlayerPlay = NULL;
+static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue = NULL;
+#if 0
+static SLVolumeItf bqPlayerVolume;
+#endif
+
+/* recorder interfaces */
+static SLObjectItf recorderObject = NULL;
+static SLRecordItf recorderRecord = NULL;
+static SLAndroidSimpleBufferQueueItf recorderBufferQueue = NULL;
+
+#if 0
+static const char *sldevaudiorecorderstr = "SLES Audio Recorder";
+static const char *sldevaudioplayerstr   = "SLES Audio Player";
+
+#define SLES_DEV_AUDIO_RECORDER sldevaudiorecorderstr
+#define SLES_DEV_AUDIO_PLAYER   sldevaudioplayerstr
+static void openslES_DetectDevices( int iscapture )
+{
+    LOGI( "openSLES_DetectDevices()" );
+    if ( iscapture )
+            addfn( SLES_DEV_AUDIO_RECORDER );
+    else
+            addfn( SLES_DEV_AUDIO_PLAYER );
+}
+#endif
+
+static void openslES_DestroyEngine(void)
+{
+    LOGI("openslES_DestroyEngine()");
+
+    /* destroy output mix object, and invalidate all associated interfaces */
+    if (outputMixObject != NULL) {
+        (*outputMixObject)->Destroy(outputMixObject);
+        outputMixObject = NULL;
+    }
+
+    /* destroy engine object, and invalidate all associated interfaces */
+    if (engineObject != NULL) {
+        (*engineObject)->Destroy(engineObject);
+        engineObject = NULL;
+        engineEngine = NULL;
+    }
+}
+
+static int openslES_CreateEngine(void)
+{
+    const SLInterfaceID ids[1] = { SL_IID_VOLUME };
+    const SLboolean req[1] = { SL_BOOLEAN_FALSE };
+    SLresult result;
+
+    LOGI("openSLES_CreateEngine()");
+
+    /* create engine */
+    result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("slCreateEngine failed: %d", result);
+        goto error;
+    }
+    LOGI("slCreateEngine OK");
+
+    /* realize the engine */
+    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeEngine failed: %d", result);
+        goto error;
+    }
+    LOGI("RealizeEngine OK");
+
+    /* get the engine interface, which is needed in order to create other objects */
+    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("EngineGetInterface failed: %d", result);
+        goto error;
+    }
+    LOGI("EngineGetInterface OK");
+
+    /* create output mix */
+    result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateOutputMix failed: %d", result);
+        goto error;
+    }
+    LOGI("CreateOutputMix OK");
+
+    /* realize the output mix */
+    result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeOutputMix failed: %d", result);
+        goto error;
+    }
+    return 1;
+
+error:
+    openslES_DestroyEngine();
+    return 0;
+}
+
+/* this callback handler is called every time a buffer finishes recording */
+static void bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
+{
+    struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
+
+    LOGV("SLES: Recording Callback");
+    SDL_SemPost(audiodata->playsem);
+}
+
+static void openslES_DestroyPCMRecorder(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+    SLresult result;
+
+    /* stop recording */
+    if (recorderRecord != NULL) {
+        result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("SetRecordState stopped: %d", result);
+        }
+    }
+
+    /* destroy audio recorder object, and invalidate all associated interfaces */
+    if (recorderObject != NULL) {
+        (*recorderObject)->Destroy(recorderObject);
+        recorderObject = NULL;
+        recorderRecord = NULL;
+        recorderBufferQueue = NULL;
+    }
+
+    if (audiodata->playsem) {
+        SDL_DestroySemaphore(audiodata->playsem);
+        audiodata->playsem = NULL;
+    }
+
+    if (audiodata->mixbuff) {
+        SDL_free(audiodata->mixbuff);
+    }
+}
+
+static int openslES_CreatePCMRecorder(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+    SLDataFormat_PCM format_pcm;
+    SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
+    SLDataSink audioSnk;
+    SLDataLocator_IODevice loc_dev;
+    SLDataSource audioSrc;
+    const SLInterfaceID ids[1] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
+    const SLboolean req[1] = { SL_BOOLEAN_TRUE };
+    SLresult result;
+    int i;
+
+    if (!Android_JNI_RequestPermission("android.permission.RECORD_AUDIO")) {
+        LOGE("This app doesn't have RECORD_AUDIO permission");
+        return SDL_SetError("This app doesn't have RECORD_AUDIO permission");
+    }
+
+    /* Just go with signed 16-bit audio as it's the most compatible */
+    this->spec.format = AUDIO_S16SYS;
+    this->spec.channels = 1;
+    /*this->spec.freq = SL_SAMPLINGRATE_16 / 1000;*/
+
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    LOGI("Try to open %u hz %u bit chan %u %s samples %u",
+         this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
+         this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
+
+    /* configure audio source */
+    loc_dev.locatorType = SL_DATALOCATOR_IODEVICE;
+    loc_dev.deviceType = SL_IODEVICE_AUDIOINPUT;
+    loc_dev.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT;
+    loc_dev.device = NULL;
+    audioSrc.pLocator = &loc_dev;
+    audioSrc.pFormat = NULL;
+
+    /* configure audio sink */
+    loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    loc_bufq.numBuffers = NUM_BUFFERS;
+
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = this->spec.channels;
+    format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
+    format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
+    format_pcm.containerSize = SDL_AUDIO_BITSIZE(this->spec.format);
+    format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
+
+    audioSnk.pLocator = &loc_bufq;
+    audioSnk.pFormat = &format_pcm;
+
+    /* create audio recorder */
+    /* (requires the RECORD_AUDIO permission) */
+    result = (*engineEngine)->CreateAudioRecorder(engineEngine, &recorderObject, &audioSrc, &audioSnk, 1, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateAudioRecorder failed: %d", result);
+        goto failed;
+    }
+
+    /* realize the recorder */
+    result = (*recorderObject)->Realize(recorderObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    /* get the record interface */
+    result = (*recorderObject)->GetInterface(recorderObject, SL_IID_RECORD, &recorderRecord);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_RECORD interface get failed: %d", result);
+        goto failed;
+    }
+
+    /* get the buffer queue interface */
+    result = (*recorderObject)->GetInterface(recorderObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &recorderBufferQueue);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_BUFFERQUEUE interface get failed: %d", result);
+        goto failed;
+    }
+
+    /* register callback on the buffer queue */
+    /* context is '(SDL_PrivateAudioData *)this->hidden' */
+    result = (*recorderBufferQueue)->RegisterCallback(recorderBufferQueue, bqRecorderCallback, this->hidden);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RegisterCallback failed: %d", result);
+        goto failed;
+    }
+
+    /* Create the audio buffer semaphore */
+    audiodata->playsem = SDL_CreateSemaphore(0);
+    if (!audiodata->playsem) {
+        LOGE("cannot create Semaphore!");
+        goto failed;
+    }
+
+    /* Create the sound buffers */
+    audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * this->spec.size);
+    if (audiodata->mixbuff == NULL) {
+        LOGE("mixbuffer allocate - out of memory");
+        goto failed;
+    }
+
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        audiodata->pmixbuff[i] = audiodata->mixbuff + i * this->spec.size;
+    }
+
+    /* in case already recording, stop recording and clear buffer queue */
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record set state failed: %d", result);
+        goto failed;
+    }
+
+    /* enqueue empty buffers to be filled by the recorder */
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, audiodata->pmixbuff[i], this->spec.size);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("Record enqueue buffers failed: %d", result);
+            goto failed;
+        }
+    }
+
+    /* start recording */
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_RECORDING);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record set state failed: %d", result);
+        goto failed;
+    }
+
+    return 0;
+
+failed:
+    return SDL_SetError("Open device failed!");
+}
+
+/* this callback handler is called every time a buffer finishes playing */
+static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
+{
+    struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
+
+    LOGV("SLES: Playback Callback");
+    SDL_SemPost(audiodata->playsem);
+}
+
+static void openslES_DestroyPCMPlayer(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+    SLresult result;
+
+    /* set the player's state to 'stopped' */
+    if (bqPlayerPlay != NULL) {
+        result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_STOPPED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("SetPlayState stopped failed: %d", result);
+        }
+    }
+
+    /* destroy buffer queue audio player object, and invalidate all associated interfaces */
+    if (bqPlayerObject != NULL) {
+        (*bqPlayerObject)->Destroy(bqPlayerObject);
+
+        bqPlayerObject = NULL;
+        bqPlayerPlay = NULL;
+        bqPlayerBufferQueue = NULL;
+    }
+
+    if (audiodata->playsem) {
+        SDL_DestroySemaphore(audiodata->playsem);
+        audiodata->playsem = NULL;
+    }
+
+    if (audiodata->mixbuff) {
+        SDL_free(audiodata->mixbuff);
+    }
+}
+
+static int openslES_CreatePCMPlayer(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+    SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
+    SLDataFormat_PCM format_pcm;
+    SLAndroidDataFormat_PCM_EX format_pcm_ex;
+    SLDataSource audioSrc;
+    SLDataSink audioSnk;
+    SLDataLocator_OutputMix loc_outmix;
+    const SLInterfaceID ids[2] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_VOLUME };
+    const SLboolean req[2] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE };
+    SLresult result;
+    int i;
+
+    /* If we want to add floating point audio support (requires API level 21)
+       it can be done as described here:
+        https://developer.android.com/ndk/guides/audio/opensl/android-extensions.html#floating-point
+    */
+    if (SDL_GetAndroidSDKVersion() >= 21) {
+        SDL_AudioFormat test_format;
+        for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+            if (SDL_AUDIO_ISSIGNED(test_format)) {
+                break;
+            }
+        }
+
+        if (!test_format) {
+            /* Didn't find a compatible format : */
+            LOGI("No compatible audio format, using signed 16-bit audio");
+            test_format = AUDIO_S16SYS;
+        }
+        this->spec.format = test_format;
+    } else {
+        /* Just go with signed 16-bit audio as it's the most compatible */
+        this->spec.format = AUDIO_S16SYS;
+    }
+
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    LOGI("Try to open %u hz %s %u bit chan %u %s samples %u",
+         this->spec.freq, SDL_AUDIO_ISFLOAT(this->spec.format) ? "float" : "pcm", SDL_AUDIO_BITSIZE(this->spec.format),
+         this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
+
+    /* configure audio source */
+    loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    loc_bufq.numBuffers = NUM_BUFFERS;
+
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = this->spec.channels;
+    format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
+    format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
+    format_pcm.containerSize = SDL_AUDIO_BITSIZE(this->spec.format);
+
+    if (SDL_AUDIO_ISBIGENDIAN(this->spec.format)) {
+        format_pcm.endianness = SL_BYTEORDER_BIGENDIAN;
+    } else {
+        format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    }
+
+    switch (this->spec.channels) {
+    case 1:
+        format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT;
+        break;
+    case 2:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO;
+        break;
+    case 3:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_FRONT_CENTER;
+        break;
+    case 4:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD;
+        break;
+    case 5:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER;
+        break;
+    case 6:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1;
+        break;
+    case 7:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_BACK_CENTER;
+        break;
+    case 8:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_7DOT1;
+        break;
+    default:
+        /* Unknown number of channels, fall back to stereo */
+        this->spec.channels = 2;
+        format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+        break;
+    }
+
+    if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
+        /* Copy all setup into PCM EX structure */
+        format_pcm_ex.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
+        format_pcm_ex.endianness = format_pcm.endianness;
+        format_pcm_ex.channelMask = format_pcm.channelMask;
+        format_pcm_ex.numChannels = format_pcm.numChannels;
+        format_pcm_ex.sampleRate = format_pcm.samplesPerSec;
+        format_pcm_ex.bitsPerSample = format_pcm.bitsPerSample;
+        format_pcm_ex.containerSize = format_pcm.containerSize;
+        format_pcm_ex.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
+    }
+
+    audioSrc.pLocator = &loc_bufq;
+    audioSrc.pFormat = SDL_AUDIO_ISFLOAT(this->spec.format) ? (void *)&format_pcm_ex : (void *)&format_pcm;
+
+    /* configure audio sink */
+    loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+    loc_outmix.outputMix = outputMixObject;
+    audioSnk.pLocator = &loc_outmix;
+    audioSnk.pFormat = NULL;
+
+    /* create audio player */
+    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    /* realize the player */
+    result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    /* get the play interface */
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_PLAY interface get failed: %d", result);
+        goto failed;
+    }
+
+    /* get the buffer queue interface */
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bqPlayerBufferQueue);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_BUFFERQUEUE interface get failed: %d", result);
+        goto failed;
+    }
+
+    /* register callback on the buffer queue */
+    /* context is '(SDL_PrivateAudioData *)this->hidden' */
+    result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, this->hidden);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RegisterCallback failed: %d", result);
+        goto failed;
+    }
+
+#if 0
+    /* get the volume interface */
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_VOLUME interface get failed: %d", result);
+        /* goto failed; */
+    }
+#endif
+
+    /* Create the audio buffer semaphore */
+    audiodata->playsem = SDL_CreateSemaphore(NUM_BUFFERS - 1);
+    if (!audiodata->playsem) {
+        LOGE("cannot create Semaphore!");
+        goto failed;
+    }
+
+    /* Create the sound buffers */
+    audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * this->spec.size);
+    if (audiodata->mixbuff == NULL) {
+        LOGE("mixbuffer allocate - out of memory");
+        goto failed;
+    }
+
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        audiodata->pmixbuff[i] = audiodata->mixbuff + i * this->spec.size;
+    }
+
+    /* set the player's state to playing */
+    result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Play set state failed: %d", result);
+        goto failed;
+    }
+
+    return 0;
+
+failed:
+    return -1;
+}
+
+static int openslES_OpenDevice(_THIS, const char *devname)
+{
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    if (this->iscapture) {
+        LOGI("openslES_OpenDevice() %s for capture", devname);
+        return openslES_CreatePCMRecorder(this);
+    } else {
+        int ret;
+        LOGI("openslES_OpenDevice() %s for playing", devname);
+        ret = openslES_CreatePCMPlayer(this);
+        if (ret < 0) {
+            /* Another attempt to open the device with a lower frequency */
+            if (this->spec.freq > 48000) {
+                openslES_DestroyPCMPlayer(this);
+                this->spec.freq = 48000;
+                ret = openslES_CreatePCMPlayer(this);
+            }
+        }
+
+        if (ret == 0) {
+            return 0;
+        } else {
+            return SDL_SetError("Open device failed!");
+        }
+    }
+}
+
+static void openslES_WaitDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+
+    LOGV("openslES_WaitDevice()");
+
+    /* Wait for an audio chunk to finish */
+    SDL_SemWait(audiodata->playsem);
+}
+
+static void openslES_PlayDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+    SLresult result;
+
+    LOGV("======openslES_PlayDevice()======");
+
+    /* Queue it up */
+    result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, audiodata->pmixbuff[audiodata->next_buffer], this->spec.size);
+
+    audiodata->next_buffer++;
+    if (audiodata->next_buffer >= NUM_BUFFERS) {
+        audiodata->next_buffer = 0;
+    }
+
+    /* If Enqueue fails, callback won't be called.
+     * Post the semphore, not to run out of buffer */
+    if (SL_RESULT_SUCCESS != result) {
+        SDL_SemPost(audiodata->playsem);
+    }
+}
+
+/*/           n   playn sem */
+/* getbuf     0   -     1 */
+/* fill buff  0   -     1 */
+/* play       0 - 0     1 */
+/* wait       1   0     0 */
+/* getbuf     1   0     0 */
+/* fill buff  1   0     0 */
+/* play       0   0     0 */
+/* wait */
+/* */
+/* okay.. */
+
+static Uint8 *openslES_GetDeviceBuf(_THIS)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+
+    LOGV("openslES_GetDeviceBuf()");
+    return audiodata->pmixbuff[audiodata->next_buffer];
+}
+
+static int openslES_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *audiodata = this->hidden;
+    SLresult result;
+
+    /* Wait for new recorded data */
+    SDL_SemWait(audiodata->playsem);
+
+    /* Copy it to the output buffer */
+    SDL_assert(buflen == this->spec.size);
+    SDL_memcpy(buffer, audiodata->pmixbuff[audiodata->next_buffer], this->spec.size);
+
+    /* Re-enqueue the buffer */
+    result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, audiodata->pmixbuff[audiodata->next_buffer], this->spec.size);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record enqueue buffers failed: %d", result);
+        return -1;
+    }
+
+    audiodata->next_buffer++;
+    if (audiodata->next_buffer >= NUM_BUFFERS) {
+        audiodata->next_buffer = 0;
+    }
+
+    return this->spec.size;
+}
+
+static void openslES_CloseDevice(_THIS)
+{
+    /* struct SDL_PrivateAudioData *audiodata = this->hidden; */
+
+    if (this->iscapture) {
+        LOGI("openslES_CloseDevice() for capture");
+        openslES_DestroyPCMRecorder(this);
+    } else {
+        LOGI("openslES_CloseDevice() for playing");
+        openslES_DestroyPCMPlayer(this);
+    }
+
+    SDL_free(this->hidden);
+}
+
+static SDL_bool openslES_Init(SDL_AudioDriverImpl *impl)
+{
+    LOGI("openslES_Init() called");
+
+    if (!openslES_CreateEngine()) {
+        return SDL_FALSE;
+    }
+
+    LOGI("openslES_Init() - set pointers");
+
+    /* Set the function pointers */
+    /* impl->DetectDevices = openslES_DetectDevices; */
+    impl->OpenDevice = openslES_OpenDevice;
+    impl->WaitDevice = openslES_WaitDevice;
+    impl->PlayDevice = openslES_PlayDevice;
+    impl->GetDeviceBuf = openslES_GetDeviceBuf;
+    impl->CaptureFromDevice = openslES_CaptureFromDevice;
+    impl->CloseDevice = openslES_CloseDevice;
+    impl->Deinitialize = openslES_DestroyEngine;
+
+    /* and the capabilities */
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+
+    LOGI("openslES_Init() - success");
+
+    /* this audio target is available. */
+    return SDL_TRUE;
+}
+
+AudioBootStrap openslES_bootstrap = {
+    "openslES", "opensl ES audio driver", openslES_Init, SDL_FALSE
+};
+
+void openslES_ResumeDevices(void)
+{
+    if (bqPlayerPlay != NULL) {
+        /* set the player's state to 'playing' */
+        SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("openslES_ResumeDevices failed: %d", result);
+        }
+    }
+}
+
+void openslES_PauseDevices(void)
+{
+    if (bqPlayerPlay != NULL) {
+        /* set the player's state to 'paused' */
+        SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PAUSED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("openslES_PauseDevices failed: %d", result);
+        }
+    }
+}
+
+#endif /* SDL_AUDIO_DRIVER_OPENSLES */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/openslES/SDL_openslES.h

@@ -0,0 +1,46 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef _SDL_openslesaudio_h
+#define _SDL_openslesaudio_h
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+#define NUM_BUFFERS 2 /* -- Don't lower this! */
+
+struct SDL_PrivateAudioData
+{
+    Uint8 *mixbuff;
+    int next_buffer;
+    Uint8 *pmixbuff[NUM_BUFFERS];
+    SDL_sem *playsem;
+};
+
+void openslES_ResumeDevices(void);
+void openslES_PauseDevices(void);
+
+#endif /* _SDL_openslesaudio_h */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/os2/SDL_os2audio.c

@@ -0,0 +1,601 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_OS2
+
+/* Allow access to a raw mixing buffer */
+
+#include "../../core/os2/SDL_os2.h"
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_os2audio.h"
+
+static PMCI_MIX_BUFFER _getNextBuffer(SDL_PrivateAudioData *pAData, PMCI_MIX_BUFFER pBuffer)
+{
+    PMCI_MIX_BUFFER pFirstBuffer = &pAData->aMixBuffers[0];
+    PMCI_MIX_BUFFER pLastBuffer = &pAData->aMixBuffers[pAData->cMixBuffers -1];
+    return (pBuffer == pLastBuffer ? pFirstBuffer : pBuffer+1);
+}
+
+static ULONG _getEnvULong(const char *name, ULONG ulMax, ULONG ulDefault)
+{
+    ULONG   ulValue;
+    char*   end;
+    char*   envval = SDL_getenv(name);
+
+    if (envval == NULL)
+        return ulDefault;
+
+    ulValue = SDL_strtoul(envval, &end, 10);
+    return (end == envval) || (ulValue > ulMax)? ulDefault : ulMax;
+}
+
+static int _MCIError(const char *func, ULONG ulResult)
+{
+    CHAR    acBuf[128];
+    mciGetErrorString(ulResult, acBuf, sizeof(acBuf));
+    return SDL_SetError("[%s] %s", func, acBuf);
+}
+
+static void _mixIOError(const char *function, ULONG ulRC)
+{
+    debug_os2("%s() - failed, rc = 0x%lX (%s)",
+              function, ulRC,
+              (ulRC == MCIERR_INVALID_MODE)   ? "Mixer mode does not match request" :
+              (ulRC == MCIERR_INVALID_BUFFER) ? "Caller sent an invalid buffer"     : "unknown");
+}
+
+static LONG APIENTRY cbAudioWriteEvent(ULONG ulStatus, PMCI_MIX_BUFFER pBuffer,
+                                       ULONG ulFlags)
+{
+    SDL_AudioDevice      *_this = (SDL_AudioDevice *)pBuffer->ulUserParm;
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    ULONG   ulRC;
+
+    debug_os2("cbAudioWriteEvent: ulStatus = %lu, pBuffer = %p, ulFlags = %#lX",ulStatus,pBuffer,ulFlags);
+
+    if (pAData->ulState == 2)
+    {
+        return 0;
+    }
+
+    if (ulFlags != MIX_WRITE_COMPLETE) {
+        debug_os2("flags = 0x%lX", ulFlags);
+        return 0;
+    }
+
+    pAData->pDrainBuffer = pBuffer;
+    ulRC = pAData->stMCIMixSetup.pmixWrite(pAData->stMCIMixSetup.ulMixHandle,
+                                           pAData->pDrainBuffer, 1);
+    if (ulRC != MCIERR_SUCCESS) {
+        _mixIOError("pmixWrite", ulRC);
+        return 0;
+    }
+
+    ulRC = DosPostEventSem(pAData->hevBuf);
+    if (ulRC != NO_ERROR && ulRC != ERROR_ALREADY_POSTED) {
+        debug_os2("DosPostEventSem(), rc = %lu", ulRC);
+    }
+
+    return 1; /* return value doesn't seem to matter. */
+}
+
+static LONG APIENTRY cbAudioReadEvent(ULONG ulStatus, PMCI_MIX_BUFFER pBuffer,
+                                      ULONG ulFlags)
+{
+    SDL_AudioDevice      *_this = (SDL_AudioDevice *)pBuffer->ulUserParm;
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    ULONG   ulRC;
+
+    debug_os2("cbAudioReadEvent: ulStatus = %lu, pBuffer = %p, ulFlags = %#lX",ulStatus,pBuffer,ulFlags);
+
+    if (pAData->ulState == 2)
+    {
+        return 0;
+    }
+
+    if (ulFlags != MIX_READ_COMPLETE) {
+        debug_os2("flags = 0x%lX", ulFlags);
+        return 0;
+    }
+
+    pAData->pFillBuffer = pBuffer;
+    if (pAData->pFillBuffer == pAData->aMixBuffers)
+    {
+       ulRC = pAData->stMCIMixSetup.pmixRead(pAData->stMCIMixSetup.ulMixHandle,
+                                             pAData->pFillBuffer, pAData->cMixBuffers);
+       if (ulRC != MCIERR_SUCCESS) {
+           _mixIOError("pmixRead", ulRC);
+           return 0;
+       }
+    }
+
+    ulRC = DosPostEventSem(pAData->hevBuf);
+    if (ulRC != NO_ERROR && ulRC != ERROR_ALREADY_POSTED) {
+        debug_os2("DosPostEventSem(), rc = %lu", ulRC);
+    }
+
+    return 1;
+}
+
+
+static void OS2_DetectDevices(void)
+{
+    MCI_SYSINFO_PARMS       stMCISysInfo;
+    CHAR                    acBuf[256];
+    ULONG                   ulDevicesNum;
+    MCI_SYSINFO_LOGDEVICE   stLogDevice;
+    MCI_SYSINFO_PARMS       stSysInfoParams;
+    ULONG                   ulRC;
+    ULONG                   ulNumber;
+    MCI_GETDEVCAPS_PARMS    stDevCapsParams;
+    MCI_OPEN_PARMS          stMCIOpen;
+    MCI_GENERIC_PARMS       stMCIGenericParams;
+
+    SDL_memset(&stMCISysInfo, 0, sizeof(stMCISysInfo));
+    acBuf[0] = '\0';
+    stMCISysInfo.pszReturn    = acBuf;
+    stMCISysInfo.ulRetSize    = sizeof(acBuf);
+    stMCISysInfo.usDeviceType = MCI_DEVTYPE_AUDIO_AMPMIX;
+    ulRC = mciSendCommand(0, MCI_SYSINFO, MCI_WAIT | MCI_SYSINFO_QUANTITY,
+                          &stMCISysInfo, 0);
+    if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+        debug_os2("MCI_SYSINFO, MCI_SYSINFO_QUANTITY - failed, rc = 0x%hX", LOUSHORT(ulRC));
+        return;
+    }
+
+    ulDevicesNum = SDL_strtoul(stMCISysInfo.pszReturn, NULL, 10);
+
+    for (ulNumber = 1; ulNumber <= ulDevicesNum;
+         ulNumber++) {
+        /* Get device install name. */
+        stSysInfoParams.ulNumber     = ulNumber;
+        stSysInfoParams.pszReturn    = acBuf;
+        stSysInfoParams.ulRetSize    = sizeof(acBuf);
+        stSysInfoParams.usDeviceType = MCI_DEVTYPE_AUDIO_AMPMIX;
+        ulRC = mciSendCommand(0, MCI_SYSINFO, MCI_WAIT | MCI_SYSINFO_INSTALLNAME,
+                              &stSysInfoParams, 0);
+        if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+            debug_os2("MCI_SYSINFO, MCI_SYSINFO_INSTALLNAME - failed, rc = 0x%hX", LOUSHORT(ulRC));
+            continue;
+        }
+
+        /* Get textual product description. */
+        stSysInfoParams.ulItem = MCI_SYSINFO_QUERY_DRIVER;
+        stSysInfoParams.pSysInfoParm = &stLogDevice;
+        SDL_strlcpy(stLogDevice.szInstallName, stSysInfoParams.pszReturn, MAX_DEVICE_NAME);
+        ulRC = mciSendCommand(0, MCI_SYSINFO, MCI_WAIT | MCI_SYSINFO_ITEM,
+                              &stSysInfoParams, 0);
+        if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+            debug_os2("MCI_SYSINFO, MCI_SYSINFO_ITEM - failed, rc = 0x%hX", LOUSHORT(ulRC));
+            continue;
+        }
+
+        SDL_AddAudioDevice(0, stLogDevice.szProductInfo, NULL, (void *)ulNumber);
+
+        /* Open audio device for querying its capabilities */
+        /* at this point we HAVE TO OPEN the waveaudio device and not the ampmix device */
+        /* because only the waveaudio device (tied to the ampmix device) supports querying for playback/record capability */
+        SDL_memset(&stMCIOpen, 0, sizeof(stMCIOpen));
+        stMCIOpen.pszDeviceType = (PSZ)MAKEULONG(MCI_DEVTYPE_WAVEFORM_AUDIO,LOUSHORT(ulNumber));
+        ulRC = mciSendCommand(0, MCI_OPEN,MCI_WAIT | MCI_OPEN_TYPE_ID | MCI_OPEN_SHAREABLE,&stMCIOpen,  0);
+        if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+            debug_os2("MCI_OPEN (getDevCaps) - failed");
+            continue;
+        }
+
+        /* check for recording capability */
+        SDL_memset(&stDevCapsParams, 0, sizeof(stDevCapsParams));
+        stDevCapsParams.ulItem = MCI_GETDEVCAPS_CAN_RECORD;
+        ulRC = mciSendCommand(stMCIOpen.usDeviceID, MCI_GETDEVCAPS, MCI_WAIT | MCI_GETDEVCAPS_ITEM,
+                              &stDevCapsParams, 0);
+        if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+            debug_os2("MCI_GETDEVCAPS, MCI_GETDEVCAPS_ITEM - failed, rc = 0x%hX", LOUSHORT(ulRC));
+        }
+        else {
+            if (stDevCapsParams.ulReturn) {
+                SDL_AddAudioDevice(1, stLogDevice.szProductInfo, NULL, (void *)(ulNumber | 0x80000000));
+            }
+        }
+
+        /* close the audio device, we are done querying its capabilities */
+        SDL_memset(&stMCIGenericParams, 0, sizeof(stMCIGenericParams));
+        ulRC = mciSendCommand(stMCIOpen.usDeviceID, MCI_CLOSE, MCI_WAIT,
+                              &stMCIGenericParams, 0);
+        if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+            debug_os2("MCI_CLOSE (getDevCaps) - failed");
+        }
+    }
+}
+
+static void OS2_WaitDevice(_THIS)
+{
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    ULONG   ulRC;
+
+    debug_os2("Enter");
+
+    /* Wait for an audio chunk to finish */
+    ulRC = DosWaitEventSem(pAData->hevBuf, 5000);
+    if (ulRC != NO_ERROR) {
+        debug_os2("DosWaitEventSem(), rc = %lu", ulRC);
+    }
+}
+
+static Uint8 *OS2_GetDeviceBuf(_THIS)
+{
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+
+    debug_os2("Enter");
+
+    return (Uint8 *) pAData->pFillBuffer->pBuffer;
+}
+
+static void OS2_PlayDevice(_THIS)
+{
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    ULONG                 ulRC;
+    PMCI_MIX_BUFFER       pMixBuffer = NULL;
+
+    debug_os2("Enter");
+
+    pMixBuffer  = pAData->pDrainBuffer;
+    pAData->pFillBuffer = _getNextBuffer(pAData, pAData->pFillBuffer);
+    if (!pAData->ulState && pAData->pFillBuffer != pMixBuffer)
+    {
+        /*
+         * this buffer was filled but we have not yet filled all buffers
+         * so just signal event sem so that OS2_WaitDevice does not need
+         * to block
+         */
+        ulRC = DosPostEventSem(pAData->hevBuf);
+    }
+
+    if (!pAData->ulState && (pAData->pFillBuffer == pMixBuffer) )
+    {
+        debug_os2("!hasStarted");
+        pAData->ulState = 1;
+
+        /* Write buffers to kick off the amp mixer */
+        ulRC = pAData->stMCIMixSetup.pmixWrite(pAData->stMCIMixSetup.ulMixHandle,
+                                               pMixBuffer, pAData->cMixBuffers);
+
+        if (ulRC != MCIERR_SUCCESS) {
+            _mixIOError("pmixWrite", ulRC);
+        }
+    }
+}
+
+static int OS2_CaptureFromDevice(_THIS,void *buffer,int buflen)
+{
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    ULONG                 ulRC;
+    PMCI_MIX_BUFFER       pMixBuffer = NULL;
+    int                   len = 0;
+
+    if (!pAData->ulState)
+    {
+        pAData->ulState = 1;
+        ulRC = pAData->stMCIMixSetup.pmixRead(pAData->stMCIMixSetup.ulMixHandle,
+                                              pAData->aMixBuffers, pAData->cMixBuffers);
+        if (ulRC != MCIERR_SUCCESS) {
+            _mixIOError("pmixRead", ulRC);
+            return -1;
+        }
+    }
+
+    /* Wait for an audio chunk to finish */
+    ulRC = DosWaitEventSem(pAData->hevBuf, 5000);
+    if (ulRC != NO_ERROR)
+    {
+        debug_os2("DosWaitEventSem(), rc = %lu", ulRC);
+        return -1;
+    }
+
+    pMixBuffer = pAData->pDrainBuffer;
+    len = SDL_min((int)pMixBuffer->ulBufferLength, buflen);
+    SDL_memcpy(buffer,pMixBuffer->pBuffer, len);
+    pAData->pDrainBuffer = _getNextBuffer(pAData, pMixBuffer);
+
+    debug_os2("buflen = %u, ulBufferLength = %lu",buflen,pMixBuffer->ulBufferLength);
+
+    return len;
+}
+
+static void OS2_FlushCapture(_THIS)
+{
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    ULONG                 ulIdx;
+
+    debug_os2("Enter");
+
+    /* Fill all device buffers with data */
+    for (ulIdx = 0; ulIdx < pAData->cMixBuffers; ulIdx++) {
+        pAData->aMixBuffers[ulIdx].ulFlags        = 0;
+        pAData->aMixBuffers[ulIdx].ulBufferLength = _this->spec.size;
+        pAData->aMixBuffers[ulIdx].ulUserParm     = (ULONG)_this;
+
+        SDL_memset(((PMCI_MIX_BUFFER)pAData->aMixBuffers)[ulIdx].pBuffer,
+                   _this->spec.silence, _this->spec.size);
+    }
+    pAData->pFillBuffer  = pAData->aMixBuffers;
+    pAData->pDrainBuffer = pAData->aMixBuffers;
+}
+
+
+static void OS2_CloseDevice(_THIS)
+{
+    SDL_PrivateAudioData *pAData = (SDL_PrivateAudioData *)_this->hidden;
+    MCI_GENERIC_PARMS     sMCIGenericParms;
+    ULONG                 ulRC;
+
+    debug_os2("Enter");
+
+    if (pAData == NULL)
+        return;
+
+    pAData->ulState = 2;
+
+    /* Close up audio */
+    if (pAData->usDeviceId != (USHORT)~0) { /* Device is open. */
+            SDL_zero(sMCIGenericParms);
+
+            ulRC = mciSendCommand(pAData->usDeviceId, MCI_STOP,
+                                  MCI_WAIT,
+                                  &sMCIGenericParms, 0);
+            if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+                debug_os2("MCI_STOP - failed" );
+            }
+
+        if (pAData->stMCIMixSetup.ulBitsPerSample != 0) { /* Mixer was initialized. */
+            ulRC = mciSendCommand(pAData->usDeviceId, MCI_MIXSETUP,
+                                  MCI_WAIT | MCI_MIXSETUP_DEINIT,
+                                  &pAData->stMCIMixSetup, 0);
+            if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+                debug_os2("MCI_MIXSETUP, MCI_MIXSETUP_DEINIT - failed");
+            }
+        }
+
+        if (pAData->cMixBuffers != 0) { /* Buffers was allocated. */
+            MCI_BUFFER_PARMS    stMCIBuffer;
+
+            stMCIBuffer.ulBufferSize = pAData->aMixBuffers[0].ulBufferLength;
+            stMCIBuffer.ulNumBuffers = pAData->cMixBuffers;
+            stMCIBuffer.pBufList = pAData->aMixBuffers;
+
+            ulRC = mciSendCommand(pAData->usDeviceId, MCI_BUFFER,
+                                  MCI_WAIT | MCI_DEALLOCATE_MEMORY, &stMCIBuffer, 0);
+            if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+                debug_os2("MCI_BUFFER, MCI_DEALLOCATE_MEMORY - failed");
+            }
+        }
+
+        ulRC = mciSendCommand(pAData->usDeviceId, MCI_CLOSE, MCI_WAIT,
+                              &sMCIGenericParms, 0);
+        if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+            debug_os2("MCI_CLOSE - failed");
+        }
+    }
+
+    if (pAData->hevBuf != NULLHANDLE)
+        DosCloseEventSem(pAData->hevBuf);
+
+    SDL_free(pAData);
+}
+
+static int OS2_OpenDevice(_THIS, const char *devname)
+{
+    SDL_PrivateAudioData *pAData;
+    SDL_AudioFormat       test_format;
+    MCI_AMP_OPEN_PARMS    stMCIAmpOpen;
+    MCI_BUFFER_PARMS      stMCIBuffer;
+    ULONG                 ulRC;
+    ULONG                 ulIdx;
+    BOOL                  new_freq;
+    ULONG                 ulHandle = (ULONG)_this->handle;
+    SDL_bool              iscapture = _this->iscapture;
+
+    new_freq = FALSE;
+    SDL_zero(stMCIAmpOpen);
+    SDL_zero(stMCIBuffer);
+
+    for (test_format = SDL_FirstAudioFormat(_this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        if (test_format == AUDIO_U8 || test_format == AUDIO_S16)
+            break;
+    }
+    if (!test_format) {
+        debug_os2("Unsupported audio format, AUDIO_S16 used");
+        test_format = AUDIO_S16;
+    }
+
+    pAData = (SDL_PrivateAudioData *) SDL_calloc(1, sizeof(struct SDL_PrivateAudioData));
+    if (pAData == NULL)
+        return SDL_OutOfMemory();
+    _this->hidden = pAData;
+
+    ulRC = DosCreateEventSem(NULL, &pAData->hevBuf, DCE_AUTORESET, TRUE);
+    if (ulRC != NO_ERROR) {
+        debug_os2("DosCreateEventSem() failed, rc = %lu", ulRC);
+        return -1;
+    }
+
+    /* Open audio device */
+    stMCIAmpOpen.usDeviceID = 0;
+    stMCIAmpOpen.pszDeviceType = (PSZ)MAKEULONG(MCI_DEVTYPE_AUDIO_AMPMIX,LOUSHORT(ulHandle));
+    ulRC = mciSendCommand(0, MCI_OPEN,
+                          (_getEnvULong("SDL_AUDIO_SHARE", 1, 0) != 0)?
+                           MCI_WAIT | MCI_OPEN_TYPE_ID | MCI_OPEN_SHAREABLE :
+                           MCI_WAIT | MCI_OPEN_TYPE_ID,
+                          &stMCIAmpOpen,  0);
+    if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+        DosCloseEventSem(pAData->hevBuf);
+        pAData->usDeviceId = (USHORT)~0;
+        return _MCIError("MCI_OPEN", ulRC);
+    }
+    pAData->usDeviceId = stMCIAmpOpen.usDeviceID;
+
+    if (iscapture) {
+        MCI_CONNECTOR_PARMS stMCIConnector;
+        MCI_AMP_SET_PARMS   stMCIAmpSet;
+        BOOL                fLineIn = _getEnvULong("SDL_AUDIO_LINEIN", 1, 0);
+
+        /* Set particular connector. */
+        SDL_zero(stMCIConnector);
+        stMCIConnector.ulConnectorType = (fLineIn)? MCI_LINE_IN_CONNECTOR :
+                                                    MCI_MICROPHONE_CONNECTOR;
+        mciSendCommand(stMCIAmpOpen.usDeviceID, MCI_CONNECTOR,
+                       MCI_WAIT | MCI_ENABLE_CONNECTOR |
+                       MCI_CONNECTOR_TYPE, &stMCIConnector, 0);
+
+        /* Disable monitor. */
+        SDL_zero(stMCIAmpSet);
+        stMCIAmpSet.ulItem = MCI_AMP_SET_MONITOR;
+        mciSendCommand(stMCIAmpOpen.usDeviceID, MCI_SET,
+                       MCI_WAIT | MCI_SET_OFF | MCI_SET_ITEM,
+                       &stMCIAmpSet, 0);
+
+        /* Set record volume. */
+        stMCIAmpSet.ulLevel = _getEnvULong("SDL_AUDIO_RECVOL", 100, 90);
+        stMCIAmpSet.ulItem  = MCI_AMP_SET_AUDIO;
+        stMCIAmpSet.ulAudio = MCI_SET_AUDIO_ALL; /* Both cnannels. */
+        stMCIAmpSet.ulValue = (fLineIn) ? MCI_LINE_IN_CONNECTOR :
+                                          MCI_MICROPHONE_CONNECTOR ;
+
+        mciSendCommand(stMCIAmpOpen.usDeviceID, MCI_SET,
+                       MCI_WAIT | MCI_SET_AUDIO | MCI_AMP_SET_GAIN,
+                       &stMCIAmpSet, 0);
+    }
+
+    _this->spec.format = test_format;
+    _this->spec.channels = _this->spec.channels > 1 ? 2 : 1;
+    if (_this->spec.freq < 8000) {
+        _this->spec.freq = 8000;
+        new_freq = TRUE;
+    } else if (_this->spec.freq > 48000) {
+        _this->spec.freq = 48000;
+        new_freq = TRUE;
+    }
+
+    /* Setup mixer. */
+    pAData->stMCIMixSetup.ulFormatTag     = MCI_WAVE_FORMAT_PCM;
+    pAData->stMCIMixSetup.ulBitsPerSample = SDL_AUDIO_BITSIZE(test_format);
+    pAData->stMCIMixSetup.ulSamplesPerSec = _this->spec.freq;
+    pAData->stMCIMixSetup.ulChannels      = _this->spec.channels;
+    pAData->stMCIMixSetup.ulDeviceType    = MCI_DEVTYPE_WAVEFORM_AUDIO;
+    if (!iscapture) {
+        pAData->stMCIMixSetup.ulFormatMode= MCI_PLAY;
+        pAData->stMCIMixSetup.pmixEvent   = cbAudioWriteEvent;
+    } else {
+        pAData->stMCIMixSetup.ulFormatMode= MCI_RECORD;
+        pAData->stMCIMixSetup.pmixEvent   = cbAudioReadEvent;
+    }
+
+    ulRC = mciSendCommand(pAData->usDeviceId, MCI_MIXSETUP,
+                          MCI_WAIT | MCI_MIXSETUP_INIT, &pAData->stMCIMixSetup, 0);
+    if (LOUSHORT(ulRC) != MCIERR_SUCCESS && _this->spec.freq > 44100) {
+        new_freq = TRUE;
+        pAData->stMCIMixSetup.ulSamplesPerSec = 44100;
+        _this->spec.freq = 44100;
+        ulRC = mciSendCommand(pAData->usDeviceId, MCI_MIXSETUP,
+                              MCI_WAIT | MCI_MIXSETUP_INIT, &pAData->stMCIMixSetup, 0);
+    }
+
+    debug_os2("Setup mixer [BPS: %lu, Freq.: %lu, Channels: %lu]: %s",
+              pAData->stMCIMixSetup.ulBitsPerSample,
+              pAData->stMCIMixSetup.ulSamplesPerSec,
+              pAData->stMCIMixSetup.ulChannels,
+              (ulRC == MCIERR_SUCCESS)? "SUCCESS" : "FAIL");
+
+    if (ulRC != MCIERR_SUCCESS) {
+        pAData->stMCIMixSetup.ulBitsPerSample = 0;
+        return _MCIError("MCI_MIXSETUP", ulRC);
+    }
+
+    if (_this->spec.samples == 0 || new_freq == TRUE) {
+    /* also see SDL_audio.c:prepare_audiospec() */
+    /* Pick a default of ~46 ms at desired frequency */
+        Uint32 samples = (_this->spec.freq / 1000) * 46;
+        Uint32 power2 = 1;
+        while (power2 < samples) {
+            power2 <<= 1;
+        }
+        _this->spec.samples = power2;
+    }
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&_this->spec);
+
+    /* Allocate memory buffers */
+    stMCIBuffer.ulBufferSize = _this->spec.size;/* (_this->spec.freq / 1000) * 100 */
+    stMCIBuffer.ulNumBuffers = NUM_BUFFERS;
+    stMCIBuffer.pBufList     = pAData->aMixBuffers;
+
+    ulRC = mciSendCommand(pAData->usDeviceId, MCI_BUFFER,
+                          MCI_WAIT | MCI_ALLOCATE_MEMORY, &stMCIBuffer, 0);
+    if (LOUSHORT(ulRC) != MCIERR_SUCCESS) {
+        return _MCIError("MCI_BUFFER", ulRC);
+    }
+    pAData->cMixBuffers = stMCIBuffer.ulNumBuffers;
+    _this->spec.size = stMCIBuffer.ulBufferSize;
+
+    debug_os2("%s, number of mix buffers: %lu",iscapture ? "capture": "play",pAData->cMixBuffers);
+
+    /* Fill all device buffers with data */
+    for (ulIdx = 0; ulIdx < stMCIBuffer.ulNumBuffers; ulIdx++) {
+        pAData->aMixBuffers[ulIdx].ulFlags        = 0;
+        pAData->aMixBuffers[ulIdx].ulBufferLength = stMCIBuffer.ulBufferSize;
+        pAData->aMixBuffers[ulIdx].ulUserParm     = (ULONG)_this;
+
+        SDL_memset(((PMCI_MIX_BUFFER)stMCIBuffer.pBufList)[ulIdx].pBuffer,
+                   _this->spec.silence, stMCIBuffer.ulBufferSize);
+    }
+    pAData->pFillBuffer  = pAData->aMixBuffers;
+    pAData->pDrainBuffer = pAData->aMixBuffers;
+
+    return 0;
+}
+
+
+static SDL_bool OS2_Init(SDL_AudioDriverImpl * impl)
+{
+    /* Set the function pointers */
+    impl->DetectDevices = OS2_DetectDevices;
+    impl->OpenDevice    = OS2_OpenDevice;
+    impl->PlayDevice    = OS2_PlayDevice;
+    impl->WaitDevice    = OS2_WaitDevice;
+    impl->GetDeviceBuf  = OS2_GetDeviceBuf;
+    impl->CloseDevice   = OS2_CloseDevice;
+    impl->CaptureFromDevice = OS2_CaptureFromDevice ;
+    impl->FlushCapture = OS2_FlushCapture;
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+
+AudioBootStrap OS2AUDIO_bootstrap = {
+    "DART", "OS/2 DART", OS2_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_OS2 */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/os2/SDL_os2audio.h

@@ -0,0 +1,55 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_os2mm_h_
+#define SDL_os2mm_h_
+
+#include "../SDL_sysaudio.h"
+
+#define  INCL_OS2MM
+#define  INCL_PM
+#define  INCL_DOS
+#define  INCL_DOSERRORS
+#include <os2.h>
+#include <os2me.h>
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *_this
+
+#define NUM_BUFFERS 3
+
+typedef struct SDL_PrivateAudioData
+{
+    USHORT              usDeviceId;
+    BYTE                _pad[2];
+    MCI_MIXSETUP_PARMS  stMCIMixSetup;
+    HEV                 hevBuf;
+    PMCI_MIX_BUFFER     pFillBuffer;
+    PMCI_MIX_BUFFER     pDrainBuffer;
+    ULONG               ulState;
+    ULONG               cMixBuffers;
+    MCI_MIX_BUFFER      aMixBuffers[NUM_BUFFERS];
+} SDL_PrivateAudioData;
+
+#endif /* SDL_os2mm_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/paudio/SDL_paudio.c

@@ -0,0 +1,485 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_PAUDIO
+
+/* Allow access to a raw mixing buffer */
+
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/time.h>
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "SDL_stdinc.h"
+#include "../SDL_audio_c.h"
+#include "../../core/unix/SDL_poll.h"
+#include "SDL_paudio.h"
+
+/* #define DEBUG_AUDIO */
+
+/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
+ * I guess nobody ever uses audio... Shame over AIX header files.  */
+#include <sys/machine.h>
+#undef BIG_ENDIAN
+#include <sys/audio.h>
+
+/* Open the audio device for playback, and don't block if busy */
+/* #define OPEN_FLAGS   (O_WRONLY|O_NONBLOCK) */
+#define OPEN_FLAGS  O_WRONLY
+
+/* Get the name of the audio device we use for output */
+
+#ifndef _PATH_DEV_DSP
+#define _PATH_DEV_DSP   "/dev/%caud%c/%c"
+#endif
+
+static char devsettings[][3] = {
+    {'p', '0', '1'}, {'p', '0', '2'}, {'p', '0', '3'}, {'p', '0', '4'},
+    {'p', '1', '1'}, {'p', '1', '2'}, {'p', '1', '3'}, {'p', '1', '4'},
+    {'p', '2', '1'}, {'p', '2', '2'}, {'p', '2', '3'}, {'p', '2', '4'},
+    {'p', '3', '1'}, {'p', '3', '2'}, {'p', '3', '3'}, {'p', '3', '4'},
+    {'b', '0', '1'}, {'b', '0', '2'}, {'b', '0', '3'}, {'b', '0', '4'},
+    {'b', '1', '1'}, {'b', '1', '2'}, {'b', '1', '3'}, {'b', '1', '4'},
+    {'b', '2', '1'}, {'b', '2', '2'}, {'b', '2', '3'}, {'b', '2', '4'},
+    {'b', '3', '1'}, {'b', '3', '2'}, {'b', '3', '3'}, {'b', '3', '4'},
+    {'\0', '\0', '\0'}
+};
+
+static int OpenUserDefinedDevice(char *path, int maxlen, int flags)
+{
+    const char *audiodev;
+    int fd;
+
+    /* Figure out what our audio device is */
+    if ((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) {
+        audiodev = SDL_getenv("AUDIODEV");
+    }
+    if (audiodev == NULL) {
+        return -1;
+    }
+    fd = open(audiodev, flags, 0);
+    if (path != NULL) {
+        SDL_strlcpy(path, audiodev, maxlen);
+        path[maxlen - 1] = '\0';
+    }
+    return fd;
+}
+
+static int OpenAudioPath(char *path, int maxlen, int flags, int classic)
+{
+    struct stat sb;
+    int cycle = 0;
+    int fd = OpenUserDefinedDevice(path, maxlen, flags);
+
+    if (fd != -1) {
+        return fd;
+    }
+
+    /* !!! FIXME: do we really need a table here? */
+    while (devsettings[cycle][0] != '\0') {
+        char audiopath[1024];
+        SDL_snprintf(audiopath, SDL_arraysize(audiopath),
+                     _PATH_DEV_DSP,
+                     devsettings[cycle][0],
+                     devsettings[cycle][1], devsettings[cycle][2]);
+
+        if (stat(audiopath, &sb) == 0) {
+            fd = open(audiopath, flags, 0);
+            if (fd >= 0) {
+                if (path != NULL) {
+                    SDL_strlcpy(path, audiopath, maxlen);
+                }
+                return fd;
+            }
+        }
+    }
+    return -1;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void PAUDIO_WaitDevice(_THIS)
+{
+    fd_set fdset;
+
+    /* See if we need to use timed audio synchronization */
+    if (this->hidden->frame_ticks) {
+        /* Use timer for general audio synchronization */
+        Sint32 ticks;
+
+        ticks = ((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
+        if (ticks > 0) {
+            SDL_Delay(ticks);
+        }
+    } else {
+        int timeoutMS;
+        audio_buffer paud_bufinfo;
+
+        if (ioctl(this->hidden->audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
+#ifdef DEBUG_AUDIO
+            fprintf(stderr, "Couldn't get audio buffer information\n");
+#endif
+            timeoutMS = 10 * 1000;
+        } else {
+            timeoutMS = paud_bufinfo.write_buf_time;
+#ifdef DEBUG_AUDIO
+            fprintf(stderr, "Waiting for write_buf_time=%d ms\n", timeoutMS);
+#endif
+        }
+
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Waiting for audio to get ready\n");
+#endif
+        if (SDL_IOReady(this->hidden->audio_fd, SDL_IOR_WRITE, timeoutMS) <= 0) {
+            /*
+             * In general we should never print to the screen,
+             * but in this case we have no other way of letting
+             * the user know what happened.
+             */
+            fprintf(stderr, "SDL: %s - Audio timeout - buggy audio driver? (disabled)\n", strerror(errno));
+            SDL_OpenedAudioDeviceDisconnected(this);
+            /* Don't try to close - may hang */
+            this->hidden->audio_fd = -1;
+#ifdef DEBUG_AUDIO
+            fprintf(stderr, "Done disabling audio\n");
+#endif
+        }
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Ready!\n");
+#endif
+    }
+}
+
+static void PAUDIO_PlayDevice(_THIS)
+{
+    int written = 0;
+    const Uint8 *mixbuf = this->hidden->mixbuf;
+    const size_t mixlen = this->hidden->mixlen;
+
+    /* Write the audio data, checking for EAGAIN on broken audio drivers */
+    do {
+        written = write(this->hidden->audio_fd, mixbuf, mixlen);
+        if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
+            SDL_Delay(1);       /* Let a little CPU time go by */
+        }
+    } while ((written < 0) &&
+             ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));
+
+    /* If timer synchronization is enabled, set the next write frame */
+    if (this->hidden->frame_ticks) {
+        this->hidden->next_frame += this->hidden->frame_ticks;
+    }
+
+    /* If we couldn't write, assume fatal error for now */
+    if (written < 0) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+}
+
+static Uint8 *PAUDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static void PAUDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->audio_fd >= 0) {
+        close(this->hidden->audio_fd);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int PAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    const char *workaround = SDL_getenv("SDL_DSP_NOSELECT");
+    char audiodev[1024];
+    const char *err = NULL;
+    int flags;
+    int bytes_per_sample;
+    SDL_AudioFormat test_format;
+    audio_init paud_init;
+    audio_buffer paud_bufinfo;
+    audio_control paud_control;
+    audio_change paud_change;
+    int fd = -1;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Open the audio device */
+    fd = OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
+    this->hidden->audio_fd = fd;
+    if (fd < 0) {
+        return SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
+    }
+
+    /*
+     * We can't set the buffer size - just ask the device for the maximum
+     * that we can have.
+     */
+    if (ioctl(fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
+        return SDL_SetError("Couldn't get audio buffer information");
+    }
+
+    if (this->spec.channels > 1)
+        this->spec.channels = 2;
+    else
+        this->spec.channels = 1;
+
+    /*
+     * Fields in the audio_init structure:
+     *
+     * Ignored by us:
+     *
+     * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
+     * paud.slot_number;         * slot number of the adapter
+     * paud.device_id;           * adapter identification number
+     *
+     * Input:
+     *
+     * paud.srate;           * the sampling rate in Hz
+     * paud.bits_per_sample; * 8, 16, 32, ...
+     * paud.bsize;           * block size for this rate
+     * paud.mode;            * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
+     * paud.channels;        * 1=mono, 2=stereo
+     * paud.flags;           * FIXED - fixed length data
+     *                       * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
+     *                       * TWOS_COMPLEMENT - 2's complement data
+     *                       * SIGNED - signed? comment seems wrong in sys/audio.h
+     *                       * BIG_ENDIAN
+     * paud.operation;       * PLAY, RECORD
+     *
+     * Output:
+     *
+     * paud.flags;           * PITCH            - pitch is supported
+     *                       * INPUT            - input is supported
+     *                       * OUTPUT           - output is supported
+     *                       * MONITOR          - monitor is supported
+     *                       * VOLUME           - volume is supported
+     *                       * VOLUME_DELAY     - volume delay is supported
+     *                       * BALANCE          - balance is supported
+     *                       * BALANCE_DELAY    - balance delay is supported
+     *                       * TREBLE           - treble control is supported
+     *                       * BASS             - bass control is supported
+     *                       * BESTFIT_PROVIDED - best fit returned
+     *                       * LOAD_CODE        - DSP load needed
+     * paud.rc;              * NO_PLAY         - DSP code can't do play requests
+     *                       * NO_RECORD       - DSP code can't do record requests
+     *                       * INVALID_REQUEST - request was invalid
+     *                       * CONFLICT        - conflict with open's flags
+     *                       * OVERLOADED      - out of DSP MIPS or memory
+     * paud.position_resolution; * smallest increment for position
+     */
+
+    paud_init.srate = this->spec.freq;
+    paud_init.mode = PCM;
+    paud_init.operation = PLAY;
+    paud_init.channels = this->spec.channels;
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        switch (test_format) {
+        case AUDIO_U8:
+            flags = TWOS_COMPLEMENT | FIXED;
+            break;
+        case AUDIO_S8:
+            flags = SIGNED | TWOS_COMPLEMENT | FIXED;
+            break;
+        case AUDIO_S16LSB:
+            flags = SIGNED | TWOS_COMPLEMENT | FIXED;
+            break;
+        case AUDIO_S16MSB:
+            flags = BIG_ENDIAN | SIGNED | TWOS_COMPLEMENT | FIXED;
+            break;
+        case AUDIO_U16LSB:
+            flags = TWOS_COMPLEMENT | FIXED;
+            break;
+        case AUDIO_U16MSB:
+            flags = BIG_ENDIAN | TWOS_COMPLEMENT | FIXED;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+    if (!test_format) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Couldn't find any hardware audio formats\n");
+#endif
+        return SDL_SetError("%s: Unsupported audio format", "paud");
+    }
+    this->spec.format = test_format;
+    paud_init.bits_per_sample = SDL_AUDIO_BITSIZE(test_format);
+    bytes_per_sample = SDL_AUDIO_BITSIZE(test_format) / 8;
+    paud_init.flags = flags;
+
+    /*
+     * We know the buffer size and the max number of subsequent writes
+     *  that can be pending. If more than one can pend, allow the application
+     *  to do something like double buffering between our write buffer and
+     *  the device's own buffer that we are filling with write() anyway.
+     *
+     * We calculate this->spec.samples like this because
+     *  SDL_CalculateAudioSpec() will give put paud_bufinfo.write_buf_cap
+     *  (or paud_bufinfo.write_buf_cap/2) into this->spec.size in return.
+     */
+    if (paud_bufinfo.request_buf_cap == 1) {
+        this->spec.samples = paud_bufinfo.write_buf_cap
+            / bytes_per_sample / this->spec.channels;
+    } else {
+        this->spec.samples = paud_bufinfo.write_buf_cap
+            / bytes_per_sample / this->spec.channels / 2;
+    }
+    paud_init.bsize = bytes_per_sample * this->spec.channels;
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /*
+     * The AIX paud device init can't modify the values of the audio_init
+     * structure that we pass to it. So we don't need any recalculation
+     * of this stuff and no reinit call as in linux dsp code.
+     *
+     * /dev/paud supports all of the encoding formats, so we don't need
+     * to do anything like reopening the device, either.
+     */
+    if (ioctl(fd, AUDIO_INIT, &paud_init) < 0) {
+        switch (paud_init.rc) {
+        case 1:
+            err = "DSP can't do play requests";
+            break;
+        case 2:
+            err = "DSP can't do record requests";
+            break;
+        case 4:
+            err = "request was invalid";
+            break;
+        case 5:
+            err = "conflict with open's flags";
+            break;
+        case 6:
+            err = "out of DSP MIPS or memory";
+            break;
+        default:
+            err = "not documented in sys/audio.h";
+            break;
+        }
+        return SDL_SetError("paud: Couldn't set audio format (%s)", err);
+    }
+
+    /* Allocate mixing buffer */
+    this->hidden->mixlen = this->spec.size;
+    this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+
+    /*
+     * Set some paramters: full volume, first speaker that we can find.
+     * Ignore the other settings for now.
+     */
+    paud_change.input = AUDIO_IGNORE;   /* the new input source */
+    paud_change.output = OUTPUT_1;      /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
+    paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
+    paud_change.volume = 0x7fffffff;    /* volume level [0-0x7fffffff] */
+    paud_change.volume_delay = AUDIO_IGNORE;    /* the new volume delay */
+    paud_change.balance = 0x3fffffff;   /* the new balance */
+    paud_change.balance_delay = AUDIO_IGNORE;   /* the new balance delay */
+    paud_change.treble = AUDIO_IGNORE;  /* the new treble state */
+    paud_change.bass = AUDIO_IGNORE;    /* the new bass state */
+    paud_change.pitch = AUDIO_IGNORE;   /* the new pitch state */
+
+    paud_control.ioctl_request = AUDIO_CHANGE;
+    paud_control.request_info = (char *) &paud_change;
+    if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Can't change audio display settings\n");
+#endif
+    }
+
+    /*
+     * Tell the device to expect data. Actual start will wait for
+     * the first write() call.
+     */
+    paud_control.ioctl_request = AUDIO_START;
+    paud_control.position = 0;
+    if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Can't start audio play\n");
+#endif
+        return SDL_SetError("Can't start audio play");
+    }
+
+    /* Check to see if we need to use SDL_IOReady() workaround */
+    if (workaround != NULL) {
+        this->hidden->frame_ticks = (float) (this->spec.samples * 1000) /
+            this->spec.freq;
+        this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
+    }
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static SDL_bool PAUDIO_Init(SDL_AudioDriverImpl * impl)
+{
+    /* !!! FIXME: not right for device enum? */
+    int fd = OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
+    if (fd < 0) {
+        SDL_SetError("PAUDIO: Couldn't open audio device");
+        return SDL_FALSE;
+    }
+    close(fd);
+
+    /* Set the function pointers */
+    impl->OpenDevice = PAUDIO_OpenDevice;
+    impl->PlayDevice = PAUDIO_PlayDevice;
+    impl->WaitDevice = PAUDIO_WaitDevice;
+    impl->GetDeviceBuf = PAUDIO_GetDeviceBuf;
+    impl->CloseDevice = PAUDIO_CloseDevice;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;       /* !!! FIXME: add device enum! */
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+AudioBootStrap PAUDIO_bootstrap = {
+    "paud", "AIX Paudio", PAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_PAUDIO */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/paudio/SDL_paudio.h

@@ -0,0 +1,48 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_paudio_h_
+#define SDL_paudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    int audio_fd;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    /* Support for audio timing using a timer, in addition to SDL_IOReady() */
+    float frame_ticks;
+    float next_frame;
+};
+#define FUDGE_TICKS 10      /* The scheduler overhead ticks per frame */
+
+#endif /* SDL_paudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/pipewire/SDL_pipewire.h

@@ -0,0 +1,47 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#ifndef SDL_pipewire_h_
+#define SDL_pipewire_h_
+
+#include "../SDL_sysaudio.h"
+#include <pipewire/pipewire.h>
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    struct pw_thread_loop *loop;
+    struct pw_stream *stream;
+    struct pw_context *context;
+    struct SDL_DataQueue *buffer;
+
+    size_t input_buffer_packet_size;
+    Sint32 stride; /* Bytes-per-frame */
+    int stream_init_status;
+};
+
+#endif /* SDL_pipewire_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/ps2/SDL_ps2audio.c

@@ -0,0 +1,177 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+/* Output audio to nowhere... */
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_ps2audio.h"
+
+#include <kernel.h>
+#include <malloc.h>
+#include <audsrv.h>
+#include <ps2_audio_driver.h>
+
+/* The tag name used by PS2 audio */
+#define PS2AUDIO_DRIVER_NAME "ps2"
+
+static int PS2AUDIO_OpenDevice(_THIS, const char *devname)
+{
+    int i, mixlen;
+    struct audsrv_fmt_t format;
+
+    this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* These are the native supported audio PS2 configs  */
+    switch (this->spec.freq) {
+    case 11025:
+    case 12000:
+    case 22050:
+    case 24000:
+    case 32000:
+    case 44100:
+    case 48000:
+        this->spec.freq = this->spec.freq;
+        break;
+    default:
+        this->spec.freq = 48000;
+        break;
+    }
+
+    this->spec.samples = 512;
+    this->spec.channels = this->spec.channels == 1 ? 1 : 2;
+    this->spec.format = this->spec.format == AUDIO_S8 ? AUDIO_S8 : AUDIO_S16;
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    format.bits = this->spec.format == AUDIO_S8 ? 8 : 16;
+    format.freq = this->spec.freq;
+    format.channels = this->spec.channels;
+
+    this->hidden->channel = audsrv_set_format(&format);
+    audsrv_set_volume(MAX_VOLUME);
+
+    if (this->hidden->channel < 0) {
+        free(this->hidden->rawbuf);
+        this->hidden->rawbuf = NULL;
+        return SDL_SetError("Couldn't reserve hardware channel");
+    }
+
+    /* Update the fragment size as size in bytes. */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate the mixing buffer.  Its size and starting address must
+       be a multiple of 64 bytes.  Our sample count is already a multiple of
+       64, so spec->size should be a multiple of 64 as well. */
+    mixlen = this->spec.size * NUM_BUFFERS;
+    this->hidden->rawbuf = (Uint8 *)memalign(64, mixlen);
+    if (this->hidden->rawbuf == NULL) {
+        return SDL_SetError("Couldn't allocate mixing buffer");
+    }
+
+    SDL_memset(this->hidden->rawbuf, 0, mixlen);
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        this->hidden->mixbufs[i] = &this->hidden->rawbuf[i * this->spec.size];
+    }
+
+    this->hidden->next_buffer = 0;
+    return 0;
+}
+
+static void PS2AUDIO_PlayDevice(_THIS)
+{
+    uint8_t *mixbuf = this->hidden->mixbufs[this->hidden->next_buffer];
+    audsrv_play_audio((char *)mixbuf, this->spec.size);
+
+    this->hidden->next_buffer = (this->hidden->next_buffer + 1) % NUM_BUFFERS;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void PS2AUDIO_WaitDevice(_THIS)
+{
+    audsrv_wait_audio(this->spec.size);
+}
+
+static Uint8 *PS2AUDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbufs[this->hidden->next_buffer];
+}
+
+static void PS2AUDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->channel >= 0) {
+        audsrv_stop_audio();
+        this->hidden->channel = -1;
+    }
+
+    if (this->hidden->rawbuf != NULL) {
+        free(this->hidden->rawbuf);
+        this->hidden->rawbuf = NULL;
+    }
+}
+
+static void PS2AUDIO_ThreadInit(_THIS)
+{
+    /* Increase the priority of this audio thread by 1 to put it
+       ahead of other SDL threads. */
+    int32_t thid;
+    ee_thread_status_t status;
+    thid = GetThreadId();
+    if (ReferThreadStatus(GetThreadId(), &status) == 0) {
+        ChangeThreadPriority(thid, status.current_priority - 1);
+    }
+}
+
+static void PS2AUDIO_Deinitialize(void)
+{
+    deinit_audio_driver();
+}
+
+static SDL_bool PS2AUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    if (init_audio_driver() < 0) {
+        return SDL_FALSE;
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = PS2AUDIO_OpenDevice;
+    impl->PlayDevice = PS2AUDIO_PlayDevice;
+    impl->WaitDevice = PS2AUDIO_WaitDevice;
+    impl->GetDeviceBuf = PS2AUDIO_GetDeviceBuf;
+    impl->CloseDevice = PS2AUDIO_CloseDevice;
+    impl->ThreadInit = PS2AUDIO_ThreadInit;
+    impl->Deinitialize = PS2AUDIO_Deinitialize;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap PS2AUDIO_bootstrap = {
+    "ps2", "PS2 audio driver", PS2AUDIO_Init, SDL_FALSE
+};
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/ps2/SDL_ps2audio.h

@@ -0,0 +1,46 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_ps2audio_h_
+#define SDL_ps2audio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+#define NUM_BUFFERS 2
+
+struct SDL_PrivateAudioData
+{
+    /* The hardware output channel. */
+    int channel;
+    /* The raw allocated mixing buffer. */
+    Uint8 *rawbuf;
+    /* Individual mixing buffers. */
+    Uint8 *mixbufs[NUM_BUFFERS];
+    /* Index of the next available mixing buffer. */
+    int next_buffer;
+};
+
+#endif /* SDL_ps2audio_h_ */
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/psp/SDL_pspaudio.c

@@ -0,0 +1,203 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_PSP
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <malloc.h> /* memalign() */
+
+#include "SDL_audio.h"
+#include "SDL_error.h"
+#include "SDL_timer.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_pspaudio.h"
+
+#include <pspaudio.h>
+#include <pspthreadman.h>
+
+/* The tag name used by PSP audio */
+#define PSPAUDIO_DRIVER_NAME "psp"
+
+static inline SDL_bool isBasicAudioConfig(const SDL_AudioSpec *spec)
+{
+    return spec->freq == 44100;
+}
+
+static int PSPAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    int format, mixlen, i;
+
+    this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* device only natively supports S16LSB */
+    this->spec.format = AUDIO_S16LSB;
+
+    /*  PSP has some limitations with the Audio. It fully supports 44.1KHz (Mono & Stereo),
+        however with frequencies differents than 44.1KHz, it just supports Stereo,
+        so a resampler must be done for these scenarios */
+    if (isBasicAudioConfig(&this->spec)) {
+        /* The sample count must be a multiple of 64. */
+        this->spec.samples = PSP_AUDIO_SAMPLE_ALIGN(this->spec.samples);
+        /* The number of channels (1 or 2). */
+        this->spec.channels = this->spec.channels == 1 ? 1 : 2;
+        format = this->spec.channels == 1 ? PSP_AUDIO_FORMAT_MONO : PSP_AUDIO_FORMAT_STEREO;
+        this->hidden->channel = sceAudioChReserve(PSP_AUDIO_NEXT_CHANNEL, this->spec.samples, format);
+    } else {
+        /* 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11050, 8000 */
+        switch (this->spec.freq) {
+        case 8000:
+        case 11025:
+        case 12000:
+        case 16000:
+        case 22050:
+        case 24000:
+        case 32000:
+        case 44100:
+        case 48000:
+            this->spec.freq = this->spec.freq;
+            break;
+        default:
+            this->spec.freq = 48000;
+            break;
+        }
+        /* The number of samples to output in one output call (min 17, max 4111). */
+        this->spec.samples = this->spec.samples < 17 ? 17 : (this->spec.samples > 4111 ? 4111 : this->spec.samples);
+        this->spec.channels = 2; /* we're forcing the hardware to stereo. */
+        this->hidden->channel = sceAudioSRCChReserve(this->spec.samples, this->spec.freq, 2);
+    }
+
+    if (this->hidden->channel < 0) {
+        free(this->hidden->rawbuf);
+        this->hidden->rawbuf = NULL;
+        return SDL_SetError("Couldn't reserve hardware channel");
+    }
+
+    /* Update the fragment size as size in bytes. */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate the mixing buffer.  Its size and starting address must
+       be a multiple of 64 bytes.  Our sample count is already a multiple of
+       64, so spec->size should be a multiple of 64 as well. */
+    mixlen = this->spec.size * NUM_BUFFERS;
+    this->hidden->rawbuf = (Uint8 *)memalign(64, mixlen);
+    if (this->hidden->rawbuf == NULL) {
+        return SDL_SetError("Couldn't allocate mixing buffer");
+    }
+
+    SDL_memset(this->hidden->rawbuf, 0, mixlen);
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        this->hidden->mixbufs[i] = &this->hidden->rawbuf[i * this->spec.size];
+    }
+
+    this->hidden->next_buffer = 0;
+    return 0;
+}
+
+static void PSPAUDIO_PlayDevice(_THIS)
+{
+    Uint8 *mixbuf = this->hidden->mixbufs[this->hidden->next_buffer];
+    if (!isBasicAudioConfig(&this->spec)) {
+        SDL_assert(this->spec.channels == 2);
+        sceAudioSRCOutputBlocking(PSP_AUDIO_VOLUME_MAX, mixbuf);
+    } else {
+        sceAudioOutputPannedBlocking(this->hidden->channel, PSP_AUDIO_VOLUME_MAX, PSP_AUDIO_VOLUME_MAX, mixbuf);
+    }
+
+    this->hidden->next_buffer = (this->hidden->next_buffer + 1) % NUM_BUFFERS;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void PSPAUDIO_WaitDevice(_THIS)
+{
+    /* Because we block when sending audio, there's no need for this function to do anything. */
+}
+
+static Uint8 *PSPAUDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbufs[this->hidden->next_buffer];
+}
+
+static void PSPAUDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->channel >= 0) {
+        if (!isBasicAudioConfig(&this->spec)) {
+            sceAudioSRCChRelease();
+        } else {
+            sceAudioChRelease(this->hidden->channel);
+        }
+        this->hidden->channel = -1;
+    }
+
+    if (this->hidden->rawbuf != NULL) {
+        free(this->hidden->rawbuf);
+        this->hidden->rawbuf = NULL;
+    }
+}
+
+static void PSPAUDIO_ThreadInit(_THIS)
+{
+    /* Increase the priority of this audio thread by 1 to put it
+       ahead of other SDL threads. */
+    SceUID thid;
+    SceKernelThreadInfo status;
+    thid = sceKernelGetThreadId();
+    status.size = sizeof(SceKernelThreadInfo);
+    if (sceKernelReferThreadStatus(thid, &status) == 0) {
+        sceKernelChangeThreadPriority(thid, status.currentPriority - 1);
+    }
+}
+
+static SDL_bool PSPAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->OpenDevice = PSPAUDIO_OpenDevice;
+    impl->PlayDevice = PSPAUDIO_PlayDevice;
+    impl->WaitDevice = PSPAUDIO_WaitDevice;
+    impl->GetDeviceBuf = PSPAUDIO_GetDeviceBuf;
+    impl->CloseDevice = PSPAUDIO_CloseDevice;
+    impl->ThreadInit = PSPAUDIO_ThreadInit;
+
+    /* PSP audio device */
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    /*
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+    */
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap PSPAUDIO_bootstrap = {
+    "psp", "PSP audio driver", PSPAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_PSP */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/psp/SDL_pspaudio.h

@@ -0,0 +1,46 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_pspaudio_h_
+#define SDL_pspaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+#define NUM_BUFFERS 2
+
+struct SDL_PrivateAudioData
+{
+    /* The hardware output channel. */
+    int channel;
+    /* The raw allocated mixing buffer. */
+    Uint8 *rawbuf;
+    /* Individual mixing buffers. */
+    Uint8 *mixbufs[NUM_BUFFERS];
+    /* Index of the next available mixing buffer. */
+    int next_buffer;
+};
+
+#endif /* SDL_pspaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/pulseaudio/SDL_pulseaudio.c

@@ -0,0 +1,918 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+/*
+  The PulseAudio target for SDL 1.3 is based on the 1.3 arts target, with
+   the appropriate parts replaced with the 1.2 PulseAudio target code. This
+   was the cleanest way to move it to 1.3. The 1.2 target was written by
+   Stéphan Kochen: stephan .a.t. kochen.nl
+*/
+#include "../../SDL_internal.h"
+#include "SDL_hints.h"
+
+#if SDL_AUDIO_DRIVER_PULSEAUDIO
+
+/* Allow access to a raw mixing buffer */
+
+#ifdef HAVE_SIGNAL_H
+#include <signal.h>
+#endif
+#include <unistd.h>
+#include <sys/types.h>
+#include <pulse/pulseaudio.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_pulseaudio.h"
+#include "SDL_loadso.h"
+#include "../../thread/SDL_systhread.h"
+
+/* should we include monitors in the device list? Set at SDL_Init time */
+static SDL_bool include_monitors = SDL_FALSE;
+
+#if (PA_API_VERSION < 12)
+/** Return non-zero if the passed state is one of the connected states */
+static SDL_INLINE int PA_CONTEXT_IS_GOOD(pa_context_state_t x)
+{
+    return x == PA_CONTEXT_CONNECTING || x == PA_CONTEXT_AUTHORIZING || x == PA_CONTEXT_SETTING_NAME || x == PA_CONTEXT_READY;
+}
+/** Return non-zero if the passed state is one of the connected states */
+static SDL_INLINE int PA_STREAM_IS_GOOD(pa_stream_state_t x)
+{
+    return x == PA_STREAM_CREATING || x == PA_STREAM_READY;
+}
+#endif /* pulseaudio <= 0.9.10 */
+
+static const char *(*PULSEAUDIO_pa_get_library_version)(void);
+static pa_channel_map *(*PULSEAUDIO_pa_channel_map_init_auto)(
+    pa_channel_map *, unsigned, pa_channel_map_def_t);
+static const char *(*PULSEAUDIO_pa_strerror)(int);
+static pa_mainloop *(*PULSEAUDIO_pa_mainloop_new)(void);
+static pa_mainloop_api *(*PULSEAUDIO_pa_mainloop_get_api)(pa_mainloop *);
+static int (*PULSEAUDIO_pa_mainloop_iterate)(pa_mainloop *, int, int *);
+static int (*PULSEAUDIO_pa_mainloop_run)(pa_mainloop *, int *);
+static void (*PULSEAUDIO_pa_mainloop_quit)(pa_mainloop *, int);
+static void (*PULSEAUDIO_pa_mainloop_free)(pa_mainloop *);
+
+static pa_operation_state_t (*PULSEAUDIO_pa_operation_get_state)(
+    const pa_operation *);
+static void (*PULSEAUDIO_pa_operation_cancel)(pa_operation *);
+static void (*PULSEAUDIO_pa_operation_unref)(pa_operation *);
+
+static pa_context *(*PULSEAUDIO_pa_context_new)(pa_mainloop_api *,
+                                                const char *);
+static int (*PULSEAUDIO_pa_context_connect)(pa_context *, const char *,
+                                            pa_context_flags_t, const pa_spawn_api *);
+static pa_operation *(*PULSEAUDIO_pa_context_get_sink_info_list)(pa_context *, pa_sink_info_cb_t, void *);
+static pa_operation *(*PULSEAUDIO_pa_context_get_source_info_list)(pa_context *, pa_source_info_cb_t, void *);
+static pa_operation *(*PULSEAUDIO_pa_context_get_sink_info_by_index)(pa_context *, uint32_t, pa_sink_info_cb_t, void *);
+static pa_operation *(*PULSEAUDIO_pa_context_get_source_info_by_index)(pa_context *, uint32_t, pa_source_info_cb_t, void *);
+static pa_context_state_t (*PULSEAUDIO_pa_context_get_state)(const pa_context *);
+static pa_operation *(*PULSEAUDIO_pa_context_subscribe)(pa_context *, pa_subscription_mask_t, pa_context_success_cb_t, void *);
+static void (*PULSEAUDIO_pa_context_set_subscribe_callback)(pa_context *, pa_context_subscribe_cb_t, void *);
+static void (*PULSEAUDIO_pa_context_disconnect)(pa_context *);
+static void (*PULSEAUDIO_pa_context_unref)(pa_context *);
+
+static pa_stream *(*PULSEAUDIO_pa_stream_new)(pa_context *, const char *,
+                                              const pa_sample_spec *, const pa_channel_map *);
+static int (*PULSEAUDIO_pa_stream_connect_playback)(pa_stream *, const char *,
+                                                    const pa_buffer_attr *, pa_stream_flags_t, const pa_cvolume *, pa_stream *);
+static int (*PULSEAUDIO_pa_stream_connect_record)(pa_stream *, const char *,
+                                                  const pa_buffer_attr *, pa_stream_flags_t);
+static pa_stream_state_t (*PULSEAUDIO_pa_stream_get_state)(const pa_stream *);
+static size_t (*PULSEAUDIO_pa_stream_writable_size)(const pa_stream *);
+static size_t (*PULSEAUDIO_pa_stream_readable_size)(const pa_stream *);
+static int (*PULSEAUDIO_pa_stream_write)(pa_stream *, const void *, size_t,
+                                         pa_free_cb_t, int64_t, pa_seek_mode_t);
+static pa_operation *(*PULSEAUDIO_pa_stream_drain)(pa_stream *,
+                                                   pa_stream_success_cb_t, void *);
+static int (*PULSEAUDIO_pa_stream_peek)(pa_stream *, const void **, size_t *);
+static int (*PULSEAUDIO_pa_stream_drop)(pa_stream *);
+static pa_operation *(*PULSEAUDIO_pa_stream_flush)(pa_stream *,
+                                                   pa_stream_success_cb_t, void *);
+static int (*PULSEAUDIO_pa_stream_disconnect)(pa_stream *);
+static void (*PULSEAUDIO_pa_stream_unref)(pa_stream *);
+static void (*PULSEAUDIO_pa_stream_set_write_callback)(pa_stream *, pa_stream_request_cb_t, void *);
+static pa_operation *(*PULSEAUDIO_pa_context_get_server_info)(pa_context *, pa_server_info_cb_t, void *);
+
+static int load_pulseaudio_syms(void);
+
+#ifdef SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC
+
+static const char *pulseaudio_library = SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC;
+static void *pulseaudio_handle = NULL;
+
+static int load_pulseaudio_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(pulseaudio_handle, fn);
+    if (*addr == NULL) {
+        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        return 0;
+    }
+
+    return 1;
+}
+
+/* cast funcs to char* first, to please GCC's strict aliasing rules. */
+#define SDL_PULSEAUDIO_SYM(x)                                       \
+    if (!load_pulseaudio_sym(#x, (void **)(char *)&PULSEAUDIO_##x)) \
+    return -1
+
+static void UnloadPulseAudioLibrary(void)
+{
+    if (pulseaudio_handle != NULL) {
+        SDL_UnloadObject(pulseaudio_handle);
+        pulseaudio_handle = NULL;
+    }
+}
+
+static int LoadPulseAudioLibrary(void)
+{
+    int retval = 0;
+    if (pulseaudio_handle == NULL) {
+        pulseaudio_handle = SDL_LoadObject(pulseaudio_library);
+        if (pulseaudio_handle == NULL) {
+            retval = -1;
+            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
+        } else {
+            retval = load_pulseaudio_syms();
+            if (retval < 0) {
+                UnloadPulseAudioLibrary();
+            }
+        }
+    }
+    return retval;
+}
+
+#else
+
+#define SDL_PULSEAUDIO_SYM(x) PULSEAUDIO_##x = x
+
+static void UnloadPulseAudioLibrary(void)
+{
+}
+
+static int LoadPulseAudioLibrary(void)
+{
+    load_pulseaudio_syms();
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC */
+
+static int load_pulseaudio_syms(void)
+{
+    SDL_PULSEAUDIO_SYM(pa_get_library_version);
+    SDL_PULSEAUDIO_SYM(pa_mainloop_new);
+    SDL_PULSEAUDIO_SYM(pa_mainloop_get_api);
+    SDL_PULSEAUDIO_SYM(pa_mainloop_iterate);
+    SDL_PULSEAUDIO_SYM(pa_mainloop_run);
+    SDL_PULSEAUDIO_SYM(pa_mainloop_quit);
+    SDL_PULSEAUDIO_SYM(pa_mainloop_free);
+    SDL_PULSEAUDIO_SYM(pa_operation_get_state);
+    SDL_PULSEAUDIO_SYM(pa_operation_cancel);
+    SDL_PULSEAUDIO_SYM(pa_operation_unref);
+    SDL_PULSEAUDIO_SYM(pa_context_new);
+    SDL_PULSEAUDIO_SYM(pa_context_connect);
+    SDL_PULSEAUDIO_SYM(pa_context_get_sink_info_list);
+    SDL_PULSEAUDIO_SYM(pa_context_get_source_info_list);
+    SDL_PULSEAUDIO_SYM(pa_context_get_sink_info_by_index);
+    SDL_PULSEAUDIO_SYM(pa_context_get_source_info_by_index);
+    SDL_PULSEAUDIO_SYM(pa_context_get_state);
+    SDL_PULSEAUDIO_SYM(pa_context_subscribe);
+    SDL_PULSEAUDIO_SYM(pa_context_set_subscribe_callback);
+    SDL_PULSEAUDIO_SYM(pa_context_disconnect);
+    SDL_PULSEAUDIO_SYM(pa_context_unref);
+    SDL_PULSEAUDIO_SYM(pa_stream_new);
+    SDL_PULSEAUDIO_SYM(pa_stream_connect_playback);
+    SDL_PULSEAUDIO_SYM(pa_stream_connect_record);
+    SDL_PULSEAUDIO_SYM(pa_stream_get_state);
+    SDL_PULSEAUDIO_SYM(pa_stream_writable_size);
+    SDL_PULSEAUDIO_SYM(pa_stream_readable_size);
+    SDL_PULSEAUDIO_SYM(pa_stream_write);
+    SDL_PULSEAUDIO_SYM(pa_stream_drain);
+    SDL_PULSEAUDIO_SYM(pa_stream_disconnect);
+    SDL_PULSEAUDIO_SYM(pa_stream_peek);
+    SDL_PULSEAUDIO_SYM(pa_stream_drop);
+    SDL_PULSEAUDIO_SYM(pa_stream_flush);
+    SDL_PULSEAUDIO_SYM(pa_stream_unref);
+    SDL_PULSEAUDIO_SYM(pa_channel_map_init_auto);
+    SDL_PULSEAUDIO_SYM(pa_strerror);
+    SDL_PULSEAUDIO_SYM(pa_stream_set_write_callback);
+    SDL_PULSEAUDIO_SYM(pa_context_get_server_info);
+    return 0;
+}
+
+static SDL_INLINE int squashVersion(const int major, const int minor, const int patch)
+{
+    return ((major & 0xFF) << 16) | ((minor & 0xFF) << 8) | (patch & 0xFF);
+}
+
+/* Workaround for older pulse: pa_context_new() must have non-NULL appname */
+static const char *getAppName(void)
+{
+    const char *retval = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_APP_NAME);
+    if (retval && *retval) {
+        return retval;
+    }
+    retval = SDL_GetHint(SDL_HINT_APP_NAME);
+    if (retval && *retval) {
+        return retval;
+    } else {
+        const char *verstr = PULSEAUDIO_pa_get_library_version();
+        retval = "SDL Application"; /* the "oh well" default. */
+        if (verstr != NULL) {
+            int maj, min, patch;
+            if (SDL_sscanf(verstr, "%d.%d.%d", &maj, &min, &patch) == 3) {
+                if (squashVersion(maj, min, patch) >= squashVersion(0, 9, 15)) {
+                    retval = NULL; /* 0.9.15+ handles NULL correctly. */
+                }
+            }
+        }
+    }
+    return retval;
+}
+
+static void WaitForPulseOperation(pa_mainloop *mainloop, pa_operation *o)
+{
+    /* This checks for NO errors currently. Either fix that, check results elsewhere, or do things you don't care about. */
+    if (mainloop && o) {
+        SDL_bool okay = SDL_TRUE;
+        while (okay && (PULSEAUDIO_pa_operation_get_state(o) == PA_OPERATION_RUNNING)) {
+            okay = (PULSEAUDIO_pa_mainloop_iterate(mainloop, 1, NULL) >= 0);
+        }
+        PULSEAUDIO_pa_operation_unref(o);
+    }
+}
+
+static void DisconnectFromPulseServer(pa_mainloop *mainloop, pa_context *context)
+{
+    if (context) {
+        PULSEAUDIO_pa_context_disconnect(context);
+        PULSEAUDIO_pa_context_unref(context);
+    }
+    if (mainloop != NULL) {
+        PULSEAUDIO_pa_mainloop_free(mainloop);
+    }
+}
+
+static int ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
+{
+    pa_mainloop *mainloop = NULL;
+    pa_context *context = NULL;
+    pa_mainloop_api *mainloop_api = NULL;
+    int state = 0;
+
+    *_mainloop = NULL;
+    *_context = NULL;
+
+    /* Set up a new main loop */
+    if (!(mainloop = PULSEAUDIO_pa_mainloop_new())) {
+        return SDL_SetError("pa_mainloop_new() failed");
+    }
+
+    mainloop_api = PULSEAUDIO_pa_mainloop_get_api(mainloop);
+    SDL_assert(mainloop_api); /* this never fails, right? */
+
+    context = PULSEAUDIO_pa_context_new(mainloop_api, getAppName());
+    if (context == NULL) {
+        PULSEAUDIO_pa_mainloop_free(mainloop);
+        return SDL_SetError("pa_context_new() failed");
+    }
+
+    /* Connect to the PulseAudio server */
+    if (PULSEAUDIO_pa_context_connect(context, NULL, 0, NULL) < 0) {
+        PULSEAUDIO_pa_context_unref(context);
+        PULSEAUDIO_pa_mainloop_free(mainloop);
+        return SDL_SetError("Could not setup connection to PulseAudio");
+    }
+
+    do {
+        if (PULSEAUDIO_pa_mainloop_iterate(mainloop, 1, NULL) < 0) {
+            PULSEAUDIO_pa_context_unref(context);
+            PULSEAUDIO_pa_mainloop_free(mainloop);
+            return SDL_SetError("pa_mainloop_iterate() failed");
+        }
+        state = PULSEAUDIO_pa_context_get_state(context);
+        if (!PA_CONTEXT_IS_GOOD(state)) {
+            PULSEAUDIO_pa_context_unref(context);
+            PULSEAUDIO_pa_mainloop_free(mainloop);
+            return SDL_SetError("Could not connect to PulseAudio");
+        }
+    } while (state != PA_CONTEXT_READY);
+
+    *_context = context;
+    *_mainloop = mainloop;
+
+    return 0; /* connected and ready! */
+}
+
+static int ConnectToPulseServer(pa_mainloop **_mainloop, pa_context **_context)
+{
+    const int retval = ConnectToPulseServer_Internal(_mainloop, _context);
+    if (retval < 0) {
+        DisconnectFromPulseServer(*_mainloop, *_context);
+    }
+    return retval;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void PULSEAUDIO_WaitDevice(_THIS)
+{
+    /* this is a no-op; we wait in PULSEAUDIO_PlayDevice now. */
+}
+
+static void WriteCallback(pa_stream *p, size_t nbytes, void *userdata)
+{
+    struct SDL_PrivateAudioData *h = (struct SDL_PrivateAudioData *)userdata;
+    /*printf("PULSEAUDIO WRITE CALLBACK! nbytes=%u\n", (unsigned int) nbytes);*/
+    h->bytes_requested += nbytes;
+}
+
+static void PULSEAUDIO_PlayDevice(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    int available = h->mixlen;
+    int written = 0;
+    int cpy;
+
+    /*printf("PULSEAUDIO PLAYDEVICE START! mixlen=%d\n", available);*/
+
+    while (SDL_AtomicGet(&this->enabled) && (available > 0)) {
+        cpy = SDL_min(h->bytes_requested, available);
+        if (cpy) {
+            if (PULSEAUDIO_pa_stream_write(h->stream, h->mixbuf + written, cpy, NULL, 0LL, PA_SEEK_RELATIVE) < 0) {
+                SDL_OpenedAudioDeviceDisconnected(this);
+                return;
+            }
+            /*printf("PULSEAUDIO FEED! nbytes=%u\n", (unsigned int) cpy);*/
+            h->bytes_requested -= cpy;
+            written += cpy;
+            available -= cpy;
+        }
+
+        /* let WriteCallback fire if necessary. */
+        /*printf("PULSEAUDIO ITERATE!\n");*/
+        if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||
+            PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
+            PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
+            SDL_OpenedAudioDeviceDisconnected(this);
+            return;
+        }
+    }
+
+    /*printf("PULSEAUDIO PLAYDEVICE END! written=%d\n", written);*/
+}
+
+static Uint8 *PULSEAUDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static int PULSEAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    const void *data = NULL;
+    size_t nbytes = 0;
+
+    while (SDL_AtomicGet(&this->enabled)) {
+        if (h->capturebuf != NULL) {
+            const int cpy = SDL_min(buflen, h->capturelen);
+            SDL_memcpy(buffer, h->capturebuf, cpy);
+            /*printf("PULSEAUDIO: fed %d captured bytes\n", cpy);*/
+            h->capturebuf += cpy;
+            h->capturelen -= cpy;
+            if (h->capturelen == 0) {
+                h->capturebuf = NULL;
+                PULSEAUDIO_pa_stream_drop(h->stream); /* done with this fragment. */
+            }
+            return cpy; /* new data, return it. */
+        }
+
+        if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||
+            PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
+            PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
+            SDL_OpenedAudioDeviceDisconnected(this);
+            return -1; /* uhoh, pulse failed! */
+        }
+
+        if (PULSEAUDIO_pa_stream_readable_size(h->stream) == 0) {
+            continue; /* no data available yet. */
+        }
+
+        /* a new fragment is available! */
+        PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
+        SDL_assert(nbytes > 0);
+        /* If data == NULL, then the buffer had a hole, ignore that */
+        if (data == NULL) {
+            PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
+        } else {
+            /* store this fragment's data, start feeding it to SDL. */
+            /*printf("PULSEAUDIO: captured %d new bytes\n", (int) nbytes);*/
+            h->capturebuf = (const Uint8 *)data;
+            h->capturelen = nbytes;
+        }
+    }
+
+    return -1; /* not enabled? */
+}
+
+static void PULSEAUDIO_FlushCapture(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    const void *data = NULL;
+    size_t nbytes = 0;
+
+    if (h->capturebuf != NULL) {
+        PULSEAUDIO_pa_stream_drop(h->stream);
+        h->capturebuf = NULL;
+        h->capturelen = 0;
+    }
+
+    while (SDL_AtomicGet(&this->enabled)) {
+        if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||
+            PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
+            PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
+            SDL_OpenedAudioDeviceDisconnected(this);
+            return; /* uhoh, pulse failed! */
+        }
+
+        if (PULSEAUDIO_pa_stream_readable_size(h->stream) == 0) {
+            break; /* no data available, so we're done. */
+        }
+
+        /* a new fragment is available! Just dump it. */
+        PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
+        PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
+    }
+}
+
+static void PULSEAUDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->stream) {
+        if (this->hidden->capturebuf != NULL) {
+            PULSEAUDIO_pa_stream_drop(this->hidden->stream);
+        }
+        PULSEAUDIO_pa_stream_disconnect(this->hidden->stream);
+        PULSEAUDIO_pa_stream_unref(this->hidden->stream);
+    }
+
+    DisconnectFromPulseServer(this->hidden->mainloop, this->hidden->context);
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden->device_name);
+    SDL_free(this->hidden);
+}
+
+static void SinkDeviceNameCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
+{
+    if (i) {
+        char **devname = (char **)data;
+        *devname = SDL_strdup(i->name);
+    }
+}
+
+static void SourceDeviceNameCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
+{
+    if (i) {
+        char **devname = (char **)data;
+        *devname = SDL_strdup(i->name);
+    }
+}
+
+static SDL_bool FindDeviceName(struct SDL_PrivateAudioData *h, const SDL_bool iscapture, void *handle)
+{
+    const uint32_t idx = ((uint32_t)((intptr_t)handle)) - 1;
+
+    if (handle == NULL) { /* NULL == default device. */
+        return SDL_TRUE;
+    }
+
+    if (iscapture) {
+        WaitForPulseOperation(h->mainloop,
+                              PULSEAUDIO_pa_context_get_source_info_by_index(h->context, idx,
+                                                                             SourceDeviceNameCallback, &h->device_name));
+    } else {
+        WaitForPulseOperation(h->mainloop,
+                              PULSEAUDIO_pa_context_get_sink_info_by_index(h->context, idx,
+                                                                           SinkDeviceNameCallback, &h->device_name));
+    }
+
+    return h->device_name != NULL;
+}
+
+static int PULSEAUDIO_OpenDevice(_THIS, const char *devname)
+{
+    struct SDL_PrivateAudioData *h = NULL;
+    SDL_AudioFormat test_format;
+    pa_sample_spec paspec;
+    pa_buffer_attr paattr;
+    pa_channel_map pacmap;
+    pa_stream_flags_t flags = 0;
+    const char *name = NULL;
+    SDL_bool iscapture = this->iscapture;
+    int state = 0, format = PA_SAMPLE_INVALID;
+    int rc = 0;
+
+    /* Initialize all variables that we clean on shutdown */
+    h = this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+#ifdef DEBUG_AUDIO
+        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
+#endif
+        switch (test_format) {
+        case AUDIO_U8:
+            format = PA_SAMPLE_U8;
+            break;
+        case AUDIO_S16LSB:
+            format = PA_SAMPLE_S16LE;
+            break;
+        case AUDIO_S16MSB:
+            format = PA_SAMPLE_S16BE;
+            break;
+        case AUDIO_S32LSB:
+            format = PA_SAMPLE_S32LE;
+            break;
+        case AUDIO_S32MSB:
+            format = PA_SAMPLE_S32BE;
+            break;
+        case AUDIO_F32LSB:
+            format = PA_SAMPLE_FLOAT32LE;
+            break;
+        case AUDIO_F32MSB:
+            format = PA_SAMPLE_FLOAT32BE;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "pulseaudio");
+    }
+    this->spec.format = test_format;
+    paspec.format = format;
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    if (!iscapture) {
+        h->mixlen = this->spec.size;
+        h->mixbuf = (Uint8 *)SDL_malloc(h->mixlen);
+        if (h->mixbuf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        SDL_memset(h->mixbuf, this->spec.silence, this->spec.size);
+    }
+
+    paspec.channels = this->spec.channels;
+    paspec.rate = this->spec.freq;
+
+    /* Reduced prebuffering compared to the defaults. */
+    paattr.fragsize = this->spec.size;
+    paattr.tlength = h->mixlen;
+    paattr.prebuf = -1;
+    paattr.maxlength = -1;
+    paattr.minreq = -1;
+    flags |= PA_STREAM_ADJUST_LATENCY;
+
+    if (ConnectToPulseServer(&h->mainloop, &h->context) < 0) {
+        return SDL_SetError("Could not connect to PulseAudio server");
+    }
+
+    if (!FindDeviceName(h, iscapture, this->handle)) {
+        return SDL_SetError("Requested PulseAudio sink/source missing?");
+    }
+
+    /* The SDL ALSA output hints us that we use Windows' channel mapping */
+    /* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
+    PULSEAUDIO_pa_channel_map_init_auto(&pacmap, this->spec.channels,
+                                        PA_CHANNEL_MAP_WAVEEX);
+
+    name = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_STREAM_NAME);
+
+    h->stream = PULSEAUDIO_pa_stream_new(
+        h->context,
+        (name && *name) ? name : "Audio Stream", /* stream description */
+        &paspec,                                 /* sample format spec */
+        &pacmap                                  /* channel map */
+    );
+
+    if (h->stream == NULL) {
+        return SDL_SetError("Could not set up PulseAudio stream");
+    }
+
+    /* now that we have multi-device support, don't move a stream from
+        a device that was unplugged to something else, unless we're default. */
+    if (h->device_name != NULL) {
+        flags |= PA_STREAM_DONT_MOVE;
+    }
+
+    if (iscapture) {
+        rc = PULSEAUDIO_pa_stream_connect_record(h->stream, h->device_name, &paattr, flags);
+    } else {
+        PULSEAUDIO_pa_stream_set_write_callback(h->stream, WriteCallback, h);
+        rc = PULSEAUDIO_pa_stream_connect_playback(h->stream, h->device_name, &paattr, flags, NULL, NULL);
+    }
+
+    if (rc < 0) {
+        return SDL_SetError("Could not connect PulseAudio stream");
+    }
+
+    do {
+        if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
+            return SDL_SetError("pa_mainloop_iterate() failed");
+        }
+        state = PULSEAUDIO_pa_stream_get_state(h->stream);
+        if (!PA_STREAM_IS_GOOD(state)) {
+            return SDL_SetError("Could not connect PulseAudio stream");
+        }
+    } while (state != PA_STREAM_READY);
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static pa_mainloop *hotplug_mainloop = NULL;
+static pa_context *hotplug_context = NULL;
+static SDL_Thread *hotplug_thread = NULL;
+
+/* These are the OS identifiers (i.e. ALSA strings)... */
+static char *default_sink_path = NULL;
+static char *default_source_path = NULL;
+/* ... and these are the descriptions we use in GetDefaultAudioInfo. */
+static char *default_sink_name = NULL;
+static char *default_source_name = NULL;
+
+/* device handles are device index + 1, cast to void*, so we never pass a NULL. */
+
+static SDL_AudioFormat PulseFormatToSDLFormat(pa_sample_format_t format)
+{
+    switch (format) {
+    case PA_SAMPLE_U8:
+        return AUDIO_U8;
+    case PA_SAMPLE_S16LE:
+        return AUDIO_S16LSB;
+    case PA_SAMPLE_S16BE:
+        return AUDIO_S16MSB;
+    case PA_SAMPLE_S32LE:
+        return AUDIO_S32LSB;
+    case PA_SAMPLE_S32BE:
+        return AUDIO_S32MSB;
+    case PA_SAMPLE_FLOAT32LE:
+        return AUDIO_F32LSB;
+    case PA_SAMPLE_FLOAT32BE:
+        return AUDIO_F32MSB;
+    default:
+        return 0;
+    }
+}
+
+/* This is called when PulseAudio adds an output ("sink") device. */
+static void SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
+{
+    SDL_AudioSpec spec;
+    SDL_bool add = (SDL_bool)((intptr_t)data);
+    if (i) {
+        spec.freq = i->sample_spec.rate;
+        spec.channels = i->sample_spec.channels;
+        spec.format = PulseFormatToSDLFormat(i->sample_spec.format);
+        spec.silence = 0;
+        spec.samples = 0;
+        spec.size = 0;
+        spec.callback = NULL;
+        spec.userdata = NULL;
+
+        if (add) {
+            SDL_AddAudioDevice(SDL_FALSE, i->description, &spec, (void *)((intptr_t)i->index + 1));
+        }
+
+        if (default_sink_path != NULL && SDL_strcmp(i->name, default_sink_path) == 0) {
+            if (default_sink_name != NULL) {
+                SDL_free(default_sink_name);
+            }
+            default_sink_name = SDL_strdup(i->description);
+        }
+    }
+}
+
+/* This is called when PulseAudio adds a capture ("source") device. */
+static void SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
+{
+    SDL_AudioSpec spec;
+    SDL_bool add = (SDL_bool)((intptr_t)data);
+    if (i) {
+        /* Maybe skip "monitor" sources. These are just output from other sinks. */
+        if (include_monitors || (i->monitor_of_sink == PA_INVALID_INDEX)) {
+            spec.freq = i->sample_spec.rate;
+            spec.channels = i->sample_spec.channels;
+            spec.format = PulseFormatToSDLFormat(i->sample_spec.format);
+            spec.silence = 0;
+            spec.samples = 0;
+            spec.size = 0;
+            spec.callback = NULL;
+            spec.userdata = NULL;
+
+            if (add) {
+                SDL_AddAudioDevice(SDL_TRUE, i->description, &spec, (void *)((intptr_t)i->index + 1));
+            }
+
+            if (default_source_path != NULL && SDL_strcmp(i->name, default_source_path) == 0) {
+                if (default_source_name != NULL) {
+                    SDL_free(default_source_name);
+                }
+                default_source_name = SDL_strdup(i->description);
+            }
+        }
+    }
+}
+
+static void ServerInfoCallback(pa_context *c, const pa_server_info *i, void *data)
+{
+    if (default_sink_path != NULL) {
+        SDL_free(default_sink_path);
+    }
+    if (default_source_path != NULL) {
+        SDL_free(default_source_path);
+    }
+    default_sink_path = SDL_strdup(i->default_sink_name);
+    default_source_path = SDL_strdup(i->default_source_name);
+}
+
+/* This is called when PulseAudio has a device connected/removed/changed. */
+static void HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *data)
+{
+    const SDL_bool added = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW);
+    const SDL_bool removed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE);
+    const SDL_bool changed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_CHANGE);
+
+    if (added || removed || changed) { /* we only care about add/remove events. */
+        const SDL_bool sink = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK);
+        const SDL_bool source = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE);
+
+        /* adds need sink details from the PulseAudio server. Another callback... */
+        if ((added || changed) && sink) {
+            if (changed) {
+                PULSEAUDIO_pa_context_get_server_info(hotplug_context, ServerInfoCallback, NULL);
+            }
+            PULSEAUDIO_pa_context_get_sink_info_by_index(hotplug_context, idx, SinkInfoCallback, (void *)((intptr_t)added));
+        } else if ((added || changed) && source) {
+            if (changed) {
+                PULSEAUDIO_pa_context_get_server_info(hotplug_context, ServerInfoCallback, NULL);
+            }
+            PULSEAUDIO_pa_context_get_source_info_by_index(hotplug_context, idx, SourceInfoCallback, (void *)((intptr_t)added));
+        } else if (removed && (sink || source)) {
+            /* removes we can handle just with the device index. */
+            SDL_RemoveAudioDevice(source != 0, (void *)((intptr_t)idx + 1));
+        }
+    }
+}
+
+/* this runs as a thread while the Pulse target is initialized to catch hotplug events. */
+static int SDLCALL HotplugThread(void *data)
+{
+    pa_operation *o;
+    SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
+    PULSEAUDIO_pa_context_set_subscribe_callback(hotplug_context, HotplugCallback, NULL);
+    o = PULSEAUDIO_pa_context_subscribe(hotplug_context, PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SOURCE, NULL, NULL);
+    PULSEAUDIO_pa_operation_unref(o); /* don't wait for it, just do our thing. */
+    PULSEAUDIO_pa_mainloop_run(hotplug_mainloop, NULL);
+    return 0;
+}
+
+static void PULSEAUDIO_DetectDevices()
+{
+    WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_server_info(hotplug_context, ServerInfoCallback, NULL));
+    WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_sink_info_list(hotplug_context, SinkInfoCallback, (void *)((intptr_t)SDL_TRUE)));
+    WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_source_info_list(hotplug_context, SourceInfoCallback, (void *)((intptr_t)SDL_TRUE)));
+
+    /* ok, we have a sane list, let's set up hotplug notifications now... */
+    hotplug_thread = SDL_CreateThreadInternal(HotplugThread, "PulseHotplug", 256 * 1024, NULL);
+}
+
+static int PULSEAUDIO_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
+{
+    int i;
+    int numdevices;
+
+    char *target;
+    if (iscapture) {
+        if (default_source_name == NULL) {
+            return SDL_SetError("PulseAudio could not find a default source");
+        }
+        target = default_source_name;
+    } else {
+        if (default_sink_name == NULL) {
+            return SDL_SetError("PulseAudio could not find a default sink");
+        }
+        target = default_sink_name;
+    }
+
+    numdevices = SDL_GetNumAudioDevices(iscapture);
+    for (i = 0; i < numdevices; i += 1) {
+        if (SDL_strcmp(SDL_GetAudioDeviceName(i, iscapture), target) == 0) {
+            if (name != NULL) {
+                *name = SDL_strdup(target);
+            }
+            SDL_GetAudioDeviceSpec(i, iscapture, spec);
+            return 0;
+        }
+    }
+    return SDL_SetError("Could not find default PulseAudio device");
+}
+
+static void PULSEAUDIO_Deinitialize(void)
+{
+    if (hotplug_thread) {
+        PULSEAUDIO_pa_mainloop_quit(hotplug_mainloop, 0);
+        SDL_WaitThread(hotplug_thread, NULL);
+        hotplug_thread = NULL;
+    }
+
+    DisconnectFromPulseServer(hotplug_mainloop, hotplug_context);
+    hotplug_mainloop = NULL;
+    hotplug_context = NULL;
+
+    if (default_sink_path != NULL) {
+        SDL_free(default_sink_path);
+        default_sink_path = NULL;
+    }
+    if (default_source_path != NULL) {
+        SDL_free(default_source_path);
+        default_source_path = NULL;
+    }
+    if (default_sink_name != NULL) {
+        SDL_free(default_sink_name);
+        default_sink_name = NULL;
+    }
+    if (default_source_name != NULL) {
+        SDL_free(default_source_name);
+        default_source_name = NULL;
+    }
+
+    UnloadPulseAudioLibrary();
+}
+
+static SDL_bool PULSEAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    if (LoadPulseAudioLibrary() < 0) {
+        return SDL_FALSE;
+    }
+
+    if (ConnectToPulseServer(&hotplug_mainloop, &hotplug_context) < 0) {
+        UnloadPulseAudioLibrary();
+        return SDL_FALSE;
+    }
+
+    include_monitors = SDL_GetHintBoolean(SDL_HINT_AUDIO_INCLUDE_MONITORS, SDL_FALSE);
+
+    /* Set the function pointers */
+    impl->DetectDevices = PULSEAUDIO_DetectDevices;
+    impl->OpenDevice = PULSEAUDIO_OpenDevice;
+    impl->PlayDevice = PULSEAUDIO_PlayDevice;
+    impl->WaitDevice = PULSEAUDIO_WaitDevice;
+    impl->GetDeviceBuf = PULSEAUDIO_GetDeviceBuf;
+    impl->CloseDevice = PULSEAUDIO_CloseDevice;
+    impl->Deinitialize = PULSEAUDIO_Deinitialize;
+    impl->CaptureFromDevice = PULSEAUDIO_CaptureFromDevice;
+    impl->FlushCapture = PULSEAUDIO_FlushCapture;
+    impl->GetDefaultAudioInfo = PULSEAUDIO_GetDefaultAudioInfo;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->SupportsNonPow2Samples = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap PULSEAUDIO_bootstrap = {
+    "pulseaudio", "PulseAudio", PULSEAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/pulseaudio/SDL_pulseaudio.h

@@ -0,0 +1,54 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_pulseaudio_h_
+#define SDL_pulseaudio_h_
+
+#include <pulse/simple.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    char *device_name;
+
+    /* pulseaudio structures */
+    pa_mainloop *mainloop;
+    pa_context *context;
+    pa_stream *stream;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    int bytes_requested; /* bytes of data the hardware wants _now_. */
+
+    const Uint8 *capturebuf;
+    int capturelen;
+};
+
+#endif /* SDL_pulseaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/qsa/SDL_qsa_audio.c

@@ -0,0 +1,618 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+/*
+ * !!! FIXME: streamline this a little by removing all the
+ * !!! FIXME:  if (capture) {} else {} sections that are identical
+ * !!! FIXME:  except for one flag.
+ */
+
+/* !!! FIXME: can this target support hotplugging? */
+/* !!! FIXME: ...does SDL2 even support QNX? */
+
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_QSA
+
+#include <errno.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <signal.h>
+#include <sys/types.h>
+#include <sys/time.h>
+#include <sched.h>
+#include <sys/select.h>
+#include <sys/neutrino.h>
+#include <sys/asoundlib.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../../core/unix/SDL_poll.h"
+#include "../SDL_audio_c.h"
+#include "SDL_qsa_audio.h"
+
+/* default channel communication parameters */
+#define DEFAULT_CPARAMS_RATE   44100
+#define DEFAULT_CPARAMS_VOICES 1
+
+#define DEFAULT_CPARAMS_FRAG_SIZE 4096
+#define DEFAULT_CPARAMS_FRAGS_MIN 1
+#define DEFAULT_CPARAMS_FRAGS_MAX 1
+
+/* List of found devices */
+#define QSA_MAX_DEVICES       32
+#define QSA_MAX_NAME_LENGTH   81+16     /* Hardcoded in QSA, can't be changed */
+
+typedef struct _QSA_Device
+{
+    char name[QSA_MAX_NAME_LENGTH];     /* Long audio device name for SDL  */
+    int cardno;
+    int deviceno;
+} QSA_Device;
+
+QSA_Device qsa_playback_device[QSA_MAX_DEVICES];
+uint32_t qsa_playback_devices;
+
+QSA_Device qsa_capture_device[QSA_MAX_DEVICES];
+uint32_t qsa_capture_devices;
+
+static int QSA_SetError(const char *fn, int status)
+{
+    return SDL_SetError("QSA: %s() failed: %s", fn, snd_strerror(status));
+}
+
+/* !!! FIXME: does this need to be here? Does the SDL version not work? */
+static void QSA_ThreadInit(_THIS)
+{
+    /* Increase default 10 priority to 25 to avoid jerky sound */
+    struct sched_param param;
+    if (SchedGet(0, 0, &param) != -1) {
+        param.sched_priority = param.sched_curpriority + 15;
+        SchedSet(0, 0, SCHED_NOCHANGE, &param);
+    }
+}
+
+/* PCM channel parameters initialize function */
+static void QSA_InitAudioParams(snd_pcm_channel_params_t * cpars)
+{
+    SDL_zerop(cpars);
+    cpars->channel = SND_PCM_CHANNEL_PLAYBACK;
+    cpars->mode = SND_PCM_MODE_BLOCK;
+    cpars->start_mode = SND_PCM_START_DATA;
+    cpars->stop_mode = SND_PCM_STOP_STOP;
+    cpars->format.format = SND_PCM_SFMT_S16_LE;
+    cpars->format.interleave = 1;
+    cpars->format.rate = DEFAULT_CPARAMS_RATE;
+    cpars->format.voices = DEFAULT_CPARAMS_VOICES;
+    cpars->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
+    cpars->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
+    cpars->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void QSA_WaitDevice(_THIS)
+{
+    int result;
+
+    /* Setup timeout for playing one fragment equal to 2 seconds          */
+    /* If timeout occurred than something wrong with hardware or driver    */
+    /* For example, Vortex 8820 audio driver stucks on second DAC because */
+    /* it doesn't exist !                                                 */
+    result = SDL_IOReady(this->hidden->audio_fd,
+                         this->iscapture ? SDL_IOR_READ : SDL_IOR_WRITE,
+                         2 * 1000);
+    switch (result) {
+    case -1:
+        SDL_SetError("QSA: SDL_IOReady() failed: %s", strerror(errno));
+        break;
+    case 0:
+        SDL_SetError("QSA: timeout on buffer waiting occurred");
+        this->hidden->timeout_on_wait = 1;
+        break;
+    default:
+        this->hidden->timeout_on_wait = 0;
+        break;
+    }
+}
+
+static void QSA_PlayDevice(_THIS)
+{
+    snd_pcm_channel_status_t cstatus;
+    int written;
+    int status;
+    int towrite;
+    void *pcmbuffer;
+
+    if (!SDL_AtomicGet(&this->enabled) || !this->hidden) {
+        return;
+    }
+
+    towrite = this->spec.size;
+    pcmbuffer = this->hidden->pcm_buf;
+
+    /* Write the audio data, checking for EAGAIN (buffer full) and underrun */
+    do {
+        written =
+            snd_pcm_plugin_write(this->hidden->audio_handle, pcmbuffer,
+                                 towrite);
+        if (written != towrite) {
+            /* Check if samples playback got stuck somewhere in hardware or in */
+            /* the audio device driver */
+            if ((errno == EAGAIN) && (written == 0)) {
+                if (this->hidden->timeout_on_wait != 0) {
+                    SDL_SetError("QSA: buffer playback timeout");
+                    return;
+                }
+            }
+
+            /* Check for errors or conditions */
+            if ((errno == EAGAIN) || (errno == EWOULDBLOCK)) {
+                /* Let a little CPU time go by and try to write again */
+                SDL_Delay(1);
+
+                /* if we wrote some data */
+                towrite -= written;
+                pcmbuffer += written * this->spec.channels;
+                continue;
+            } else {
+                if ((errno == EINVAL) || (errno == EIO)) {
+                    SDL_zero(cstatus);
+                    if (!this->iscapture) {
+                        cstatus.channel = SND_PCM_CHANNEL_PLAYBACK;
+                    } else {
+                        cstatus.channel = SND_PCM_CHANNEL_CAPTURE;
+                    }
+
+                    status =
+                        snd_pcm_plugin_status(this->hidden->audio_handle,
+                                              &cstatus);
+                    if (status < 0) {
+                        QSA_SetError("snd_pcm_plugin_status", status);
+                        return;
+                    }
+
+                    if ((cstatus.status == SND_PCM_STATUS_UNDERRUN) ||
+                        (cstatus.status == SND_PCM_STATUS_READY)) {
+                        if (!this->iscapture) {
+                            status =
+                                snd_pcm_plugin_prepare(this->hidden->
+                                                       audio_handle,
+                                                       SND_PCM_CHANNEL_PLAYBACK);
+                        } else {
+                            status =
+                                snd_pcm_plugin_prepare(this->hidden->
+                                                       audio_handle,
+                                                       SND_PCM_CHANNEL_CAPTURE);
+                        }
+                        if (status < 0) {
+                            QSA_SetError("snd_pcm_plugin_prepare", status);
+                            return;
+                        }
+                    }
+                    continue;
+                } else {
+                    return;
+                }
+            }
+        } else {
+            /* we wrote all remaining data */
+            towrite -= written;
+            pcmbuffer += written * this->spec.channels;
+        }
+    } while ((towrite > 0) && SDL_AtomicGet(&this->enabled));
+
+    /* If we couldn't write, assume fatal error for now */
+    if (towrite != 0) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+}
+
+static Uint8 *QSA_GetDeviceBuf(_THIS)
+{
+    return this->hidden->pcm_buf;
+}
+
+static void QSA_CloseDevice(_THIS)
+{
+    if (this->hidden->audio_handle != NULL) {
+#if _NTO_VERSION < 710
+        if (!this->iscapture) {
+            /* Finish playing available samples */
+            snd_pcm_plugin_flush(this->hidden->audio_handle,
+                                 SND_PCM_CHANNEL_PLAYBACK);
+        } else {
+            /* Cancel unread samples during capture */
+            snd_pcm_plugin_flush(this->hidden->audio_handle,
+                                 SND_PCM_CHANNEL_CAPTURE);
+        }
+#endif
+        snd_pcm_close(this->hidden->audio_handle);
+    }
+
+    SDL_free(this->hidden->pcm_buf);
+    SDL_free(this->hidden);
+}
+
+static int QSA_OpenDevice(_THIS, const char *devname)
+{
+    const QSA_Device *device = (const QSA_Device *) this->handle;
+    SDL_bool iscapture = this->iscapture;
+    int status = 0;
+    int format = 0;
+    SDL_AudioFormat test_format;
+    snd_pcm_channel_setup_t csetup;
+    snd_pcm_channel_params_t cparams;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden =
+        (struct SDL_PrivateAudioData *) SDL_calloc(1,
+                                                   (sizeof
+                                                    (struct
+                                                     SDL_PrivateAudioData)));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    /* Initialize channel transfer parameters to default */
+    QSA_InitAudioParams(&cparams);
+
+    if (device != NULL) {
+        /* Open requested audio device */
+        this->hidden->deviceno = device->deviceno;
+        this->hidden->cardno = device->cardno;
+        status = snd_pcm_open(&this->hidden->audio_handle,
+                              device->cardno, device->deviceno,
+                              iscapture ? SND_PCM_OPEN_CAPTURE : SND_PCM_OPEN_PLAYBACK);
+    } else {
+        /* Open system default audio device */
+        status = snd_pcm_open_preferred(&this->hidden->audio_handle,
+                                        &this->hidden->cardno,
+                                        &this->hidden->deviceno,
+                                        iscapture ? SND_PCM_OPEN_CAPTURE : SND_PCM_OPEN_PLAYBACK);
+    }
+
+    /* Check if requested device is opened */
+    if (status < 0) {
+        this->hidden->audio_handle = NULL;
+        return QSA_SetError("snd_pcm_open", status);
+    }
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        /* if match found set format to equivalent QSA format */
+        switch (test_format) {
+        case AUDIO_U8:
+            format = SND_PCM_SFMT_U8;
+            break;
+        case AUDIO_S8:
+            format = SND_PCM_SFMT_S8;
+            break;
+        case AUDIO_S16LSB:
+            format = SND_PCM_SFMT_S16_LE;
+            break;
+        case AUDIO_S16MSB:
+            format = SND_PCM_SFMT_S16_BE;
+            break;
+        case AUDIO_U16LSB:
+            format = SND_PCM_SFMT_U16_LE;
+            break;
+        case AUDIO_U16MSB:
+            format = SND_PCM_SFMT_U16_BE;
+            break;
+        case AUDIO_S32LSB:
+            format = SND_PCM_SFMT_S32_LE;
+            break;
+        case AUDIO_S32MSB:
+            format = SND_PCM_SFMT_S32_BE;
+            break;
+        case AUDIO_F32LSB:
+            format = SND_PCM_SFMT_FLOAT_LE;
+            break;
+        case AUDIO_F32MSB:
+            format = SND_PCM_SFMT_FLOAT_BE;
+            break;
+        default:
+            continue;
+        }
+        break;
+    }
+    /* assumes test_format not 0 on success */
+    /* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "qsa");
+    }
+    this->spec.format = test_format;
+
+    /* Set the audio format */
+    cparams.format.format = format;
+
+    /* Set mono/stereo/4ch/6ch/8ch audio */
+    cparams.format.voices = this->spec.channels;
+
+    /* Set rate */
+    cparams.format.rate = this->spec.freq;
+
+    /* Setup the transfer parameters according to cparams */
+    status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
+    if (status < 0) {
+        return QSA_SetError("snd_pcm_plugin_params", status);
+    }
+
+    /* Make sure channel is setup right one last time */
+    SDL_zero(csetup);
+    if (!this->iscapture) {
+        csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
+    } else {
+        csetup.channel = SND_PCM_CHANNEL_CAPTURE;
+    }
+
+    /* Setup an audio channel */
+    if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) {
+        return SDL_SetError("QSA: Unable to setup channel");
+    }
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    this->hidden->pcm_len = this->spec.size;
+
+    if (this->hidden->pcm_len == 0) {
+        this->hidden->pcm_len =
+            csetup.buf.block.frag_size * this->spec.channels *
+            (snd_pcm_format_width(format) / 8);
+    }
+
+    /*
+     * Allocate memory to the audio buffer and initialize with silence
+     *  (Note that buffer size must be a multiple of fragment size, so find
+     *  closest multiple)
+     */
+    this->hidden->pcm_buf =
+        (Uint8 *) SDL_malloc(this->hidden->pcm_len);
+    if (this->hidden->pcm_buf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->pcm_buf, this->spec.silence,
+               this->hidden->pcm_len);
+
+    /* get the file descriptor */
+    if (!this->iscapture) {
+        this->hidden->audio_fd =
+            snd_pcm_file_descriptor(this->hidden->audio_handle,
+                                    SND_PCM_CHANNEL_PLAYBACK);
+    } else {
+        this->hidden->audio_fd =
+            snd_pcm_file_descriptor(this->hidden->audio_handle,
+                                    SND_PCM_CHANNEL_CAPTURE);
+    }
+
+    if (this->hidden->audio_fd < 0) {
+        return QSA_SetError("snd_pcm_file_descriptor", status);
+    }
+
+    /* Prepare an audio channel */
+    if (!this->iscapture) {
+        /* Prepare audio playback */
+        status =
+            snd_pcm_plugin_prepare(this->hidden->audio_handle,
+                                   SND_PCM_CHANNEL_PLAYBACK);
+    } else {
+        /* Prepare audio capture */
+        status =
+            snd_pcm_plugin_prepare(this->hidden->audio_handle,
+                                   SND_PCM_CHANNEL_CAPTURE);
+    }
+
+    if (status < 0) {
+        return QSA_SetError("snd_pcm_plugin_prepare", status);
+    }
+
+    /* We're really ready to rock and roll. :-) */
+    return 0;
+}
+
+static void QSA_DetectDevices(void)
+{
+    uint32_t it;
+    uint32_t cards;
+    uint32_t devices;
+    int32_t status;
+
+    /* Detect amount of available devices       */
+    /* this value can be changed in the runtime */
+    cards = snd_cards();
+
+    /* If io-audio manager is not running we will get 0 as number */
+    /* of available audio devices                                 */
+    if (cards == 0) {
+        /* We have no any available audio devices */
+        return;
+    }
+
+    /* !!! FIXME: code duplication */
+    /* Find requested devices by type */
+    {  /* output devices */
+        /* Playback devices enumeration requested */
+        for (it = 0; it < cards; it++) {
+            devices = 0;
+            do {
+                status =
+                    snd_card_get_longname(it,
+                                          qsa_playback_device
+                                          [qsa_playback_devices].name,
+                                          QSA_MAX_NAME_LENGTH);
+                if (status == EOK) {
+                    snd_pcm_t *handle;
+
+                    /* Add device number to device name */
+                    sprintf(qsa_playback_device[qsa_playback_devices].name +
+                            SDL_strlen(qsa_playback_device
+                                       [qsa_playback_devices].name), " d%d",
+                            devices);
+
+                    /* Store associated card number id */
+                    qsa_playback_device[qsa_playback_devices].cardno = it;
+
+                    /* Check if this device id could play anything */
+                    status =
+                        snd_pcm_open(&handle, it, devices,
+                                     SND_PCM_OPEN_PLAYBACK);
+                    if (status == EOK) {
+                        qsa_playback_device[qsa_playback_devices].deviceno =
+                            devices;
+                        status = snd_pcm_close(handle);
+                        if (status == EOK) {
+                            /* Note that spec is NULL, because we are required to open the device before
+                             * acquiring the mix format, making this information inaccessible at
+                             * enumeration time
+                             */
+                            SDL_AddAudioDevice(SDL_FALSE, qsa_playback_device[qsa_playback_devices].name, NULL, &qsa_playback_device[qsa_playback_devices]);
+                            qsa_playback_devices++;
+                        }
+                    } else {
+                        /* Check if we got end of devices list */
+                        if (status == -ENOENT) {
+                            break;
+                        }
+                    }
+                } else {
+                    break;
+                }
+
+                /* Check if we reached maximum devices count */
+                if (qsa_playback_devices >= QSA_MAX_DEVICES) {
+                    break;
+                }
+                devices++;
+            } while (1);
+
+            /* Check if we reached maximum devices count */
+            if (qsa_playback_devices >= QSA_MAX_DEVICES) {
+                break;
+            }
+        }
+    }
+
+    {  /* capture devices */
+        /* Capture devices enumeration requested */
+        for (it = 0; it < cards; it++) {
+            devices = 0;
+            do {
+                status =
+                    snd_card_get_longname(it,
+                                          qsa_capture_device
+                                          [qsa_capture_devices].name,
+                                          QSA_MAX_NAME_LENGTH);
+                if (status == EOK) {
+                    snd_pcm_t *handle;
+
+                    /* Add device number to device name */
+                    sprintf(qsa_capture_device[qsa_capture_devices].name +
+                            SDL_strlen(qsa_capture_device
+                                       [qsa_capture_devices].name), " d%d",
+                            devices);
+
+                    /* Store associated card number id */
+                    qsa_capture_device[qsa_capture_devices].cardno = it;
+
+                    /* Check if this device id could play anything */
+                    status =
+                        snd_pcm_open(&handle, it, devices,
+                                     SND_PCM_OPEN_CAPTURE);
+                    if (status == EOK) {
+                        qsa_capture_device[qsa_capture_devices].deviceno =
+                            devices;
+                        status = snd_pcm_close(handle);
+                        if (status == EOK) {
+                            /* Note that spec is NULL, because we are required to open the device before
+                             * acquiring the mix format, making this information inaccessible at
+                             * enumeration time
+                             */
+                            SDL_AddAudioDevice(SDL_TRUE, qsa_capture_device[qsa_capture_devices].name, NULL, &qsa_capture_device[qsa_capture_devices]);
+                            qsa_capture_devices++;
+                        }
+                    } else {
+                        /* Check if we got end of devices list */
+                        if (status == -ENOENT) {
+                            break;
+                        }
+                    }
+
+                    /* Check if we reached maximum devices count */
+                    if (qsa_capture_devices >= QSA_MAX_DEVICES) {
+                        break;
+                    }
+                } else {
+                    break;
+                }
+                devices++;
+            } while (1);
+
+            /* Check if we reached maximum devices count */
+            if (qsa_capture_devices >= QSA_MAX_DEVICES) {
+                break;
+            }
+        }
+    }
+}
+
+static void QSA_Deinitialize(void)
+{
+    /* Clear devices array on shutdown */
+    /* !!! FIXME: we zero these on init...any reason to do it here? */
+    SDL_zeroa(qsa_playback_device);
+    SDL_zeroa(qsa_capture_device);
+    qsa_playback_devices = 0;
+    qsa_capture_devices = 0;
+}
+
+static SDL_bool QSA_Init(SDL_AudioDriverImpl * impl)
+{
+    /* Clear devices array */
+    SDL_zeroa(qsa_playback_device);
+    SDL_zeroa(qsa_capture_device);
+    qsa_playback_devices = 0;
+    qsa_capture_devices = 0;
+
+    /* Set function pointers                                     */
+    /* DeviceLock and DeviceUnlock functions are used default,   */
+    /* provided by SDL, which uses pthread_mutex for lock/unlock */
+    impl->DetectDevices = QSA_DetectDevices;
+    impl->OpenDevice = QSA_OpenDevice;
+    impl->ThreadInit = QSA_ThreadInit;
+    impl->WaitDevice = QSA_WaitDevice;
+    impl->PlayDevice = QSA_PlayDevice;
+    impl->GetDeviceBuf = QSA_GetDeviceBuf;
+    impl->CloseDevice = QSA_CloseDevice;
+    impl->Deinitialize = QSA_Deinitialize;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+AudioBootStrap QSAAUDIO_bootstrap = {
+    "qsa", "QNX QSA Audio", QSA_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_QSA */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/qsa/SDL_qsa_audio.h

@@ -0,0 +1,54 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#ifndef __SDL_QSA_AUDIO_H__
+#define __SDL_QSA_AUDIO_H__
+
+#include <sys/asoundlib.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice* this
+
+struct SDL_PrivateAudioData
+{
+    /* The audio device handle */
+    int cardno;
+    int deviceno;
+    snd_pcm_t *audio_handle;
+
+    /* The audio file descriptor */
+    int audio_fd;
+
+    /* Select timeout status */
+    uint32_t timeout_on_wait;
+
+    /* Raw mixing buffer */
+    Uint8 *pcm_buf;
+    Uint32 pcm_len;
+};
+
+#endif /* __SDL_QSA_AUDIO_H__ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/sndio/SDL_sndioaudio.c

@@ -0,0 +1,371 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_SNDIO
+
+/* OpenBSD sndio target */
+
+#if HAVE_STDIO_H
+#include <stdio.h>
+#endif
+
+#ifdef HAVE_SIGNAL_H
+#include <signal.h>
+#endif
+
+#include <poll.h>
+#include <unistd.h>
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_sndioaudio.h"
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+#include "SDL_loadso.h"
+#endif
+
+#ifndef INFTIM
+#define INFTIM -1
+#endif
+
+#ifndef SIO_DEVANY
+#define SIO_DEVANY "default"
+#endif
+
+static struct sio_hdl *(*SNDIO_sio_open)(const char *, unsigned int, int);
+static void (*SNDIO_sio_close)(struct sio_hdl *);
+static int (*SNDIO_sio_setpar)(struct sio_hdl *, struct sio_par *);
+static int (*SNDIO_sio_getpar)(struct sio_hdl *, struct sio_par *);
+static int (*SNDIO_sio_start)(struct sio_hdl *);
+static int (*SNDIO_sio_stop)(struct sio_hdl *);
+static size_t (*SNDIO_sio_read)(struct sio_hdl *, void *, size_t);
+static size_t (*SNDIO_sio_write)(struct sio_hdl *, const void *, size_t);
+static int (*SNDIO_sio_nfds)(struct sio_hdl *);
+static int (*SNDIO_sio_pollfd)(struct sio_hdl *, struct pollfd *, int);
+static int (*SNDIO_sio_revents)(struct sio_hdl *, struct pollfd *);
+static int (*SNDIO_sio_eof)(struct sio_hdl *);
+static void (*SNDIO_sio_initpar)(struct sio_par *);
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+static const char *sndio_library = SDL_AUDIO_DRIVER_SNDIO_DYNAMIC;
+static void *sndio_handle = NULL;
+
+static int load_sndio_sym(const char *fn, void **addr)
+{
+    *addr = SDL_LoadFunction(sndio_handle, fn);
+    if (*addr == NULL) {
+        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        return 0;
+    }
+
+    return 1;
+}
+
+/* cast funcs to char* first, to please GCC's strict aliasing rules. */
+#define SDL_SNDIO_SYM(x)                                  \
+    if (!load_sndio_sym(#x, (void **)(char *)&SNDIO_##x)) \
+    return -1
+#else
+#define SDL_SNDIO_SYM(x) SNDIO_##x = x
+#endif
+
+static int load_sndio_syms(void)
+{
+    SDL_SNDIO_SYM(sio_open);
+    SDL_SNDIO_SYM(sio_close);
+    SDL_SNDIO_SYM(sio_setpar);
+    SDL_SNDIO_SYM(sio_getpar);
+    SDL_SNDIO_SYM(sio_start);
+    SDL_SNDIO_SYM(sio_stop);
+    SDL_SNDIO_SYM(sio_read);
+    SDL_SNDIO_SYM(sio_write);
+    SDL_SNDIO_SYM(sio_nfds);
+    SDL_SNDIO_SYM(sio_pollfd);
+    SDL_SNDIO_SYM(sio_revents);
+    SDL_SNDIO_SYM(sio_eof);
+    SDL_SNDIO_SYM(sio_initpar);
+    return 0;
+}
+
+#undef SDL_SNDIO_SYM
+
+#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
+
+static void UnloadSNDIOLibrary(void)
+{
+    if (sndio_handle != NULL) {
+        SDL_UnloadObject(sndio_handle);
+        sndio_handle = NULL;
+    }
+}
+
+static int LoadSNDIOLibrary(void)
+{
+    int retval = 0;
+    if (sndio_handle == NULL) {
+        sndio_handle = SDL_LoadObject(sndio_library);
+        if (sndio_handle == NULL) {
+            retval = -1;
+            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
+        } else {
+            retval = load_sndio_syms();
+            if (retval < 0) {
+                UnloadSNDIOLibrary();
+            }
+        }
+    }
+    return retval;
+}
+
+#else
+
+static void UnloadSNDIOLibrary(void)
+{
+}
+
+static int LoadSNDIOLibrary(void)
+{
+    load_sndio_syms();
+    return 0;
+}
+
+#endif /* SDL_AUDIO_DRIVER_SNDIO_DYNAMIC */
+
+static void SNDIO_WaitDevice(_THIS)
+{
+    /* no-op; SNDIO_sio_write() blocks if necessary. */
+}
+
+static void SNDIO_PlayDevice(_THIS)
+{
+    const int written = SNDIO_sio_write(this->hidden->dev,
+                                        this->hidden->mixbuf,
+                                        this->hidden->mixlen);
+
+    /* If we couldn't write, assume fatal error for now */
+    if (written == 0) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
+#endif
+}
+
+static int SNDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    size_t r;
+    int revents;
+    int nfds;
+
+    /* Emulate a blocking read */
+    r = SNDIO_sio_read(this->hidden->dev, buffer, buflen);
+    while (r == 0 && !SNDIO_sio_eof(this->hidden->dev)) {
+        nfds = SNDIO_sio_pollfd(this->hidden->dev, this->hidden->pfd, POLLIN);
+        if (nfds <= 0 || poll(this->hidden->pfd, nfds, INFTIM) < 0) {
+            return -1;
+        }
+        revents = SNDIO_sio_revents(this->hidden->dev, this->hidden->pfd);
+        if (revents & POLLIN) {
+            r = SNDIO_sio_read(this->hidden->dev, buffer, buflen);
+        }
+        if (revents & POLLHUP) {
+            break;
+        }
+    }
+    return (int)r;
+}
+
+static void SNDIO_FlushCapture(_THIS)
+{
+    char buf[512];
+
+    while (SNDIO_sio_read(this->hidden->dev, buf, sizeof(buf)) != 0) {
+        /* do nothing */;
+    }
+}
+
+static Uint8 *SNDIO_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbuf;
+}
+
+static void SNDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->pfd != NULL) {
+        SDL_free(this->hidden->pfd);
+    }
+    if (this->hidden->dev != NULL) {
+        SNDIO_sio_stop(this->hidden->dev);
+        SNDIO_sio_close(this->hidden->dev);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int SNDIO_OpenDevice(_THIS, const char *devname)
+{
+    SDL_AudioFormat test_format;
+    struct sio_par par;
+    SDL_bool iscapture = this->iscapture;
+
+    this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    this->hidden->mixlen = this->spec.size;
+
+    /* Capture devices must be non-blocking for SNDIO_FlushCapture */
+    this->hidden->dev = SNDIO_sio_open(devname != NULL ? devname : SIO_DEVANY,
+                                       iscapture ? SIO_REC : SIO_PLAY, iscapture);
+    if (this->hidden->dev == NULL) {
+        return SDL_SetError("sio_open() failed");
+    }
+
+    /* Allocate the pollfd array for capture devices */
+    if (iscapture) {
+        this->hidden->pfd = SDL_malloc(sizeof(struct pollfd) * SNDIO_sio_nfds(this->hidden->dev));
+        if (this->hidden->pfd == NULL) {
+            return SDL_OutOfMemory();
+        }
+    }
+
+    SNDIO_sio_initpar(&par);
+
+    par.rate = this->spec.freq;
+    par.pchan = this->spec.channels;
+    par.round = this->spec.samples;
+    par.appbufsz = par.round * 2;
+
+    /* Try for a closest match on audio format */
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        if (!SDL_AUDIO_ISFLOAT(test_format)) {
+            par.le = SDL_AUDIO_ISLITTLEENDIAN(test_format) ? 1 : 0;
+            par.sig = SDL_AUDIO_ISSIGNED(test_format) ? 1 : 0;
+            par.bits = SDL_AUDIO_BITSIZE(test_format);
+
+            if (SNDIO_sio_setpar(this->hidden->dev, &par) == 0) {
+                continue;
+            }
+            if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
+                return SDL_SetError("sio_getpar() failed");
+            }
+            if (par.bps != SIO_BPS(par.bits)) {
+                continue;
+            }
+            if ((par.bits == 8 * par.bps) || (par.msb)) {
+                break;
+            }
+        }
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "sndio");
+    }
+
+    if ((par.bps == 4) && (par.sig) && (par.le)) {
+        this->spec.format = AUDIO_S32LSB;
+    } else if ((par.bps == 4) && (par.sig) && (!par.le)) {
+        this->spec.format = AUDIO_S32MSB;
+    } else if ((par.bps == 2) && (par.sig) && (par.le)) {
+        this->spec.format = AUDIO_S16LSB;
+    } else if ((par.bps == 2) && (par.sig) && (!par.le)) {
+        this->spec.format = AUDIO_S16MSB;
+    } else if ((par.bps == 2) && (!par.sig) && (par.le)) {
+        this->spec.format = AUDIO_U16LSB;
+    } else if ((par.bps == 2) && (!par.sig) && (!par.le)) {
+        this->spec.format = AUDIO_U16MSB;
+    } else if ((par.bps == 1) && (par.sig)) {
+        this->spec.format = AUDIO_S8;
+    } else if ((par.bps == 1) && (!par.sig)) {
+        this->spec.format = AUDIO_U8;
+    } else {
+        return SDL_SetError("sndio: Got unsupported hardware audio format.");
+    }
+
+    this->spec.freq = par.rate;
+    this->spec.channels = par.pchan;
+    this->spec.samples = par.round;
+
+    /* Calculate the final parameters for this audio specification */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    this->hidden->mixlen = this->spec.size;
+    this->hidden->mixbuf = (Uint8 *)SDL_malloc(this->hidden->mixlen);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
+
+    if (!SNDIO_sio_start(this->hidden->dev)) {
+        return SDL_SetError("sio_start() failed");
+    }
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+static void SNDIO_Deinitialize(void)
+{
+    UnloadSNDIOLibrary();
+}
+
+static void SNDIO_DetectDevices(void)
+{
+    SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, NULL, (void *)0x1);
+    SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, NULL, (void *)0x2);
+}
+
+static SDL_bool SNDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    if (LoadSNDIOLibrary() < 0) {
+        return SDL_FALSE;
+    }
+
+    /* Set the function pointers */
+    impl->OpenDevice = SNDIO_OpenDevice;
+    impl->WaitDevice = SNDIO_WaitDevice;
+    impl->PlayDevice = SNDIO_PlayDevice;
+    impl->GetDeviceBuf = SNDIO_GetDeviceBuf;
+    impl->CloseDevice = SNDIO_CloseDevice;
+    impl->CaptureFromDevice = SNDIO_CaptureFromDevice;
+    impl->FlushCapture = SNDIO_FlushCapture;
+    impl->Deinitialize = SNDIO_Deinitialize;
+    impl->DetectDevices = SNDIO_DetectDevices;
+
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap SNDIO_bootstrap = {
+    "sndio", "OpenBSD sndio", SNDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_SNDIO */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/sndio/SDL_sndioaudio.h

@@ -0,0 +1,49 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_sndioaudio_h_
+#define SDL_sndioaudio_h_
+
+#include <poll.h>
+#include <sndio.h>
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The audio device handle */
+    struct sio_hdl *dev;
+
+    /* Raw mixing buffer */
+    Uint8 *mixbuf;
+    int mixlen;
+
+    /* Polling structures for non-blocking sndio devices */
+    struct pollfd *pfd;
+};
+
+#endif /* SDL_sndioaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/sun/SDL_sunaudio.c

@@ -0,0 +1,410 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_SUNAUDIO
+
+/* Allow access to a raw mixing buffer */
+
+#include <fcntl.h>
+#include <errno.h>
+#ifdef __NETBSD__
+#include <sys/ioctl.h>
+#include <sys/audioio.h>
+#endif
+#ifdef __SVR4
+#include <sys/audioio.h>
+#else
+#include <sys/time.h>
+#include <sys/types.h>
+#endif
+#include <unistd.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../../core/unix/SDL_poll.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "SDL_sunaudio.h"
+
+/* Open the audio device for playback, and don't block if busy */
+
+#if defined(AUDIO_GETINFO) && !defined(AUDIO_GETBUFINFO)
+#define AUDIO_GETBUFINFO AUDIO_GETINFO
+#endif
+
+/* Audio driver functions */
+static Uint8 snd2au(int sample);
+
+/* Audio driver bootstrap functions */
+static void SUNAUDIO_DetectDevices(void)
+{
+    SDL_EnumUnixAudioDevices(1, (int (*)(int)) NULL);
+}
+
+#ifdef DEBUG_AUDIO
+void CheckUnderflow(_THIS)
+{
+#ifdef AUDIO_GETBUFINFO
+    audio_info_t info;
+    int left;
+
+    ioctl(this->hidden->audio_fd, AUDIO_GETBUFINFO, &info);
+    left = (this->hidden->written - info.play.samples);
+    if (this->hidden->written && (left == 0)) {
+        fprintf(stderr, "audio underflow!\n");
+    }
+#endif
+}
+#endif
+
+static void SUNAUDIO_WaitDevice(_THIS)
+{
+#ifdef AUDIO_GETBUFINFO
+#define SLEEP_FUDGE 10      /* 10 ms scheduling fudge factor */
+    audio_info_t info;
+    Sint32 left;
+
+    ioctl(this->hidden->audio_fd, AUDIO_GETBUFINFO, &info);
+    left = (this->hidden->written - info.play.samples);
+    if (left > this->hidden->fragsize) {
+        Sint32 sleepy;
+
+        sleepy = ((left - this->hidden->fragsize) / this->hidden->frequency);
+        sleepy -= SLEEP_FUDGE;
+        if (sleepy > 0) {
+            SDL_Delay(sleepy);
+        }
+    }
+#else
+    SDL_IOReady(this->hidden->audio_fd, SDL_IOR_WRITE, -1);
+#endif
+}
+
+static void SUNAUDIO_PlayDevice(_THIS)
+{
+    /* Write the audio data */
+    if (this->hidden->ulaw_only) {
+        /* Assuming that this->spec.freq >= 8000 Hz */
+        int accum, incr, pos;
+        Uint8 *aubuf;
+
+        accum = 0;
+        incr = this->spec.freq / 8;
+        aubuf = this->hidden->ulaw_buf;
+        switch (this->hidden->audio_fmt & 0xFF) {
+        case 8:
+            {
+                Uint8 *sndbuf;
+
+                sndbuf = this->hidden->mixbuf;
+                for (pos = 0; pos < this->hidden->fragsize; ++pos) {
+                    *aubuf = snd2au((0x80 - *sndbuf) * 64);
+                    accum += incr;
+                    while (accum > 0) {
+                        accum -= 1000;
+                        sndbuf += 1;
+                    }
+                    aubuf += 1;
+                }
+            }
+            break;
+        case 16:
+            {
+                Sint16 *sndbuf;
+
+                sndbuf = (Sint16 *) this->hidden->mixbuf;
+                for (pos = 0; pos < this->hidden->fragsize; ++pos) {
+                    *aubuf = snd2au(*sndbuf / 4);
+                    accum += incr;
+                    while (accum > 0) {
+                        accum -= 1000;
+                        sndbuf += 1;
+                    }
+                    aubuf += 1;
+                }
+            }
+            break;
+        }
+#ifdef DEBUG_AUDIO
+        CheckUnderflow(this);
+#endif
+        if (write(this->hidden->audio_fd, this->hidden->ulaw_buf,
+            this->hidden->fragsize) < 0) {
+            /* Assume fatal error, for now */
+            SDL_OpenedAudioDeviceDisconnected(this);
+        }
+        this->hidden->written += this->hidden->fragsize;
+    } else {
+#ifdef DEBUG_AUDIO
+        CheckUnderflow(this);
+#endif
+        if (write(this->hidden->audio_fd, this->hidden->mixbuf,
+            this->spec.size) < 0) {
+            /* Assume fatal error, for now */
+            SDL_OpenedAudioDeviceDisconnected(this);
+        }
+        this->hidden->written += this->hidden->fragsize;
+    }
+}
+
+static Uint8 *SUNAUDIO_GetDeviceBuf(_THIS)
+{
+    return (this->hidden->mixbuf);
+}
+
+static void SUNAUDIO_CloseDevice(_THIS)
+{
+    SDL_free(this->hidden->ulaw_buf);
+    if (this->hidden->audio_fd >= 0) {
+        close(this->hidden->audio_fd);
+    }
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static int SUNAUDIO_OpenDevice(_THIS, const char *devname)
+{
+#ifdef AUDIO_SETINFO
+    int enc;
+#endif
+    SDL_bool iscapture = this->iscapture;
+    int desired_freq = 0;
+    const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
+    SDL_AudioFormat format = 0;
+    audio_info_t info;
+
+    /* We don't care what the devname is...we'll try to open anything. */
+    /*  ...but default to first name in the list... */
+    if (devname == NULL) {
+        devname = SDL_GetAudioDeviceName(0, iscapture);
+        if (devname == NULL) {
+            return SDL_SetError("No such audio device");
+        }
+    }
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Open the audio device */
+    this->hidden->audio_fd = open(devname, flags, 0);
+    if (this->hidden->audio_fd < 0) {
+        return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
+    }
+
+    desired_freq = this->spec.freq;
+
+    /* Determine the audio parameters from the AudioSpec */
+    switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
+
+    case 8:
+        {                       /* Unsigned 8 bit audio data */
+            this->spec.format = AUDIO_U8;
+#ifdef AUDIO_SETINFO
+            enc = AUDIO_ENCODING_LINEAR8;
+#endif
+        }
+        break;
+
+    case 16:
+        {                       /* Signed 16 bit audio data */
+            this->spec.format = AUDIO_S16SYS;
+#ifdef AUDIO_SETINFO
+            enc = AUDIO_ENCODING_LINEAR;
+#endif
+        }
+        break;
+
+    default:
+        {
+            /* !!! FIXME: fallback to conversion on unsupported types! */
+            return SDL_SetError("Unsupported audio format");
+        }
+    }
+    this->hidden->audio_fmt = this->spec.format;
+
+    this->hidden->ulaw_only = 0;    /* modern Suns do support linear audio */
+#ifdef AUDIO_SETINFO
+    for (;;) {
+        audio_info_t info;
+        AUDIO_INITINFO(&info);  /* init all fields to "no change" */
+
+        /* Try to set the requested settings */
+        info.play.sample_rate = this->spec.freq;
+        info.play.channels = this->spec.channels;
+        info.play.precision = (enc == AUDIO_ENCODING_ULAW)
+            ? 8 : this->spec.format & 0xff;
+        info.play.encoding = enc;
+        if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {
+
+            /* Check to be sure we got what we wanted */
+            if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
+                return SDL_SetError("Error getting audio parameters: %s",
+                                    strerror(errno));
+            }
+            if (info.play.encoding == enc
+                && info.play.precision == (this->spec.format & 0xff)
+                && info.play.channels == this->spec.channels) {
+                /* Yow! All seems to be well! */
+                this->spec.freq = info.play.sample_rate;
+                break;
+            }
+        }
+
+        switch (enc) {
+        case AUDIO_ENCODING_LINEAR8:
+            /* unsigned 8bit apparently not supported here */
+            enc = AUDIO_ENCODING_LINEAR;
+            this->spec.format = AUDIO_S16SYS;
+            break;              /* try again */
+
+        case AUDIO_ENCODING_LINEAR:
+            /* linear 16bit didn't work either, resort to µ-law */
+            enc = AUDIO_ENCODING_ULAW;
+            this->spec.channels = 1;
+            this->spec.freq = 8000;
+            this->spec.format = AUDIO_U8;
+            this->hidden->ulaw_only = 1;
+            break;
+
+        default:
+            /* oh well... */
+            return SDL_SetError("Error setting audio parameters: %s",
+                                strerror(errno));
+        }
+    }
+#endif /* AUDIO_SETINFO */
+    this->hidden->written = 0;
+
+    /* We can actually convert on-the-fly to U-Law */
+    if (this->hidden->ulaw_only) {
+        this->spec.freq = desired_freq;
+        this->hidden->fragsize = (this->spec.samples * 1000) /
+            (this->spec.freq / 8);
+        this->hidden->frequency = 8;
+        this->hidden->ulaw_buf = (Uint8 *) SDL_malloc(this->hidden->fragsize);
+        if (this->hidden->ulaw_buf == NULL) {
+            return SDL_OutOfMemory();
+        }
+        this->spec.channels = 1;
+    } else {
+        this->hidden->fragsize = this->spec.samples;
+        this->hidden->frequency = this->spec.freq / 1000;
+    }
+#ifdef DEBUG_AUDIO
+    fprintf(stderr, "Audio device %s U-Law only\n",
+            this->hidden->ulaw_only ? "is" : "is not");
+    fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
+            this->spec.format, this->spec.channels, this->spec.freq);
+#endif
+
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate mixing buffer */
+    this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->spec.size);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
+
+    /* We're ready to rock and roll. :-) */
+    return 0;
+}
+
+/************************************************************************/
+/* This function (snd2au()) copyrighted:                                */
+/************************************************************************/
+/*      Copyright 1989 by Rich Gopstein and Harris Corporation          */
+/*                                                                      */
+/*      Permission to use, copy, modify, and distribute this software   */
+/*      and its documentation for any purpose and without fee is        */
+/*      hereby granted, provided that the above copyright notice        */
+/*      appears in all copies and that both that copyright notice and   */
+/*      this permission notice appear in supporting documentation, and  */
+/*      that the name of Rich Gopstein and Harris Corporation not be    */
+/*      used in advertising or publicity pertaining to distribution     */
+/*      of the software without specific, written prior permission.     */
+/*      Rich Gopstein and Harris Corporation make no representations    */
+/*      about the suitability of this software for any purpose.  It     */
+/*      provided "as is" without express or implied warranty.           */
+/************************************************************************/
+
+static Uint8 snd2au(int sample)
+{
+
+    int mask;
+
+    if (sample < 0) {
+        sample = -sample;
+        mask = 0x7f;
+    } else {
+        mask = 0xff;
+    }
+
+    if (sample < 32) {
+        sample = 0xF0 | (15 - sample / 2);
+    } else if (sample < 96) {
+        sample = 0xE0 | (15 - (sample - 32) / 4);
+    } else if (sample < 224) {
+        sample = 0xD0 | (15 - (sample - 96) / 8);
+    } else if (sample < 480) {
+        sample = 0xC0 | (15 - (sample - 224) / 16);
+    } else if (sample < 992) {
+        sample = 0xB0 | (15 - (sample - 480) / 32);
+    } else if (sample < 2016) {
+        sample = 0xA0 | (15 - (sample - 992) / 64);
+    } else if (sample < 4064) {
+        sample = 0x90 | (15 - (sample - 2016) / 128);
+    } else if (sample < 8160) {
+        sample = 0x80 | (15 - (sample - 4064) / 256);
+    } else {
+        sample = 0x80;
+    }
+    return (mask & sample);
+}
+
+static SDL_bool SUNAUDIO_Init(SDL_AudioDriverImpl * impl)
+{
+    /* Set the function pointers */
+    impl->DetectDevices = SUNAUDIO_DetectDevices;
+    impl->OpenDevice = SUNAUDIO_OpenDevice;
+    impl->PlayDevice = SUNAUDIO_PlayDevice;
+    impl->WaitDevice = SUNAUDIO_WaitDevice;
+    impl->GetDeviceBuf = SUNAUDIO_GetDeviceBuf;
+    impl->CloseDevice = SUNAUDIO_CloseDevice;
+
+    impl->AllowsArbitraryDeviceNames = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap SUNAUDIO_bootstrap = {
+    "audio", "UNIX /dev/audio interface", SUNAUDIO_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_SUNAUDIO */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/sun/SDL_sunaudio.h

@@ -0,0 +1,47 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_sunaudio_h_
+#define SDL_sunaudio_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData
+{
+    /* The file descriptor for the audio device */
+    int audio_fd;
+
+    SDL_AudioFormat audio_fmt;  /* The app audio format */
+    Uint8 *mixbuf;              /* The app mixing buffer */
+    int ulaw_only;              /* Flag -- does hardware only output U-law? */
+    Uint8 *ulaw_buf;            /* The U-law mixing buffer */
+    Sint32 written;             /* The number of samples written */
+    int fragsize;               /* The audio fragment size in samples */
+    int frequency;              /* The audio frequency in KHz */
+};
+
+#endif /* SDL_sunaudio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/vita/SDL_vitaaudio.c

@@ -0,0 +1,221 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_VITA
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <malloc.h> /* memalign() */
+
+#include "SDL_audio.h"
+#include "SDL_error.h"
+#include "SDL_timer.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_vitaaudio.h"
+
+#include <psp2/kernel/threadmgr.h>
+#include <psp2/audioout.h>
+#include <psp2/audioin.h>
+
+#define SCE_AUDIO_SAMPLE_ALIGN(s) (((s) + 63) & ~63)
+#define SCE_AUDIO_MAX_VOLUME      0x8000
+
+static int VITAAUD_OpenCaptureDevice(_THIS)
+{
+    this->spec.freq = 16000;
+    this->spec.samples = 512;
+    this->spec.channels = 1;
+
+    SDL_CalculateAudioSpec(&this->spec);
+
+    this->hidden->port = sceAudioInOpenPort(SCE_AUDIO_IN_PORT_TYPE_VOICE, 512, 16000, SCE_AUDIO_IN_PARAM_FORMAT_S16_MONO);
+
+    if (this->hidden->port < 0) {
+        return SDL_SetError("Couldn't open audio in port: %x", this->hidden->port);
+    }
+
+    return 0;
+}
+
+static int VITAAUD_OpenDevice(_THIS, const char *devname)
+{
+    int format, mixlen, i, port = SCE_AUDIO_OUT_PORT_TYPE_MAIN;
+    int vols[2] = { SCE_AUDIO_MAX_VOLUME, SCE_AUDIO_MAX_VOLUME };
+    SDL_AudioFormat test_format;
+
+    this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_memset(this->hidden, 0, sizeof(*this->hidden));
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        if (test_format == AUDIO_S16LSB) {
+            this->spec.format = test_format;
+            break;
+        }
+    }
+
+    if (!test_format) {
+        return SDL_SetError("Unsupported audio format");
+    }
+
+    if (this->iscapture) {
+        return VITAAUD_OpenCaptureDevice(this);
+    }
+
+    /* The sample count must be a multiple of 64. */
+    this->spec.samples = SCE_AUDIO_SAMPLE_ALIGN(this->spec.samples);
+
+    /* Update the fragment size as size in bytes. */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Allocate the mixing buffer.  Its size and starting address must
+       be a multiple of 64 bytes.  Our sample count is already a multiple of
+       64, so spec->size should be a multiple of 64 as well. */
+    mixlen = this->spec.size * NUM_BUFFERS;
+    this->hidden->rawbuf = (Uint8 *)memalign(64, mixlen);
+    if (this->hidden->rawbuf == NULL) {
+        return SDL_SetError("Couldn't allocate mixing buffer");
+    }
+
+    /* Setup the hardware channel. */
+    if (this->spec.channels == 1) {
+        format = SCE_AUDIO_OUT_MODE_MONO;
+    } else {
+        format = SCE_AUDIO_OUT_MODE_STEREO;
+    }
+
+    if (this->spec.freq < 48000) {
+        port = SCE_AUDIO_OUT_PORT_TYPE_BGM;
+    }
+
+    this->hidden->port = sceAudioOutOpenPort(port, this->spec.samples, this->spec.freq, format);
+    if (this->hidden->port < 0) {
+        free(this->hidden->rawbuf);
+        this->hidden->rawbuf = NULL;
+        return SDL_SetError("Couldn't open audio out port: %x", this->hidden->port);
+    }
+
+    sceAudioOutSetVolume(this->hidden->port, SCE_AUDIO_VOLUME_FLAG_L_CH | SCE_AUDIO_VOLUME_FLAG_R_CH, vols);
+
+    SDL_memset(this->hidden->rawbuf, 0, mixlen);
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        this->hidden->mixbufs[i] = &this->hidden->rawbuf[i * this->spec.size];
+    }
+
+    this->hidden->next_buffer = 0;
+    return 0;
+}
+
+static void VITAAUD_PlayDevice(_THIS)
+{
+    Uint8 *mixbuf = this->hidden->mixbufs[this->hidden->next_buffer];
+
+    sceAudioOutOutput(this->hidden->port, mixbuf);
+
+    this->hidden->next_buffer = (this->hidden->next_buffer + 1) % NUM_BUFFERS;
+}
+
+/* This function waits until it is possible to write a full sound buffer */
+static void VITAAUD_WaitDevice(_THIS)
+{
+    /* Because we block when sending audio, there's no need for this function to do anything. */
+}
+
+static Uint8 *VITAAUD_GetDeviceBuf(_THIS)
+{
+    return this->hidden->mixbufs[this->hidden->next_buffer];
+}
+
+static void VITAAUD_CloseDevice(_THIS)
+{
+    if (this->hidden->port >= 0) {
+        if (this->iscapture) {
+            sceAudioInReleasePort(this->hidden->port);
+        } else {
+            sceAudioOutReleasePort(this->hidden->port);
+        }
+        this->hidden->port = -1;
+    }
+
+    if (!this->iscapture && this->hidden->rawbuf != NULL) {
+        free(this->hidden->rawbuf); /* this uses memalign(), not SDL_malloc(). */
+        this->hidden->rawbuf = NULL;
+    }
+}
+
+static int VITAAUD_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    int ret;
+    SDL_assert(buflen == this->spec.size);
+    ret = sceAudioInInput(this->hidden->port, buffer);
+    if (ret < 0) {
+        return SDL_SetError("Failed to capture from device: %x", ret);
+    }
+    return this->spec.size;
+}
+
+static void VITAAUD_ThreadInit(_THIS)
+{
+    /* Increase the priority of this audio thread by 1 to put it
+       ahead of other SDL threads. */
+    SceUID thid;
+    SceKernelThreadInfo info;
+    thid = sceKernelGetThreadId();
+    info.size = sizeof(SceKernelThreadInfo);
+    if (sceKernelGetThreadInfo(thid, &info) == 0) {
+        sceKernelChangeThreadPriority(thid, info.currentPriority - 1);
+    }
+}
+
+static SDL_bool VITAAUD_Init(SDL_AudioDriverImpl *impl)
+{
+    /* Set the function pointers */
+    impl->OpenDevice = VITAAUD_OpenDevice;
+    impl->PlayDevice = VITAAUD_PlayDevice;
+    impl->WaitDevice = VITAAUD_WaitDevice;
+    impl->GetDeviceBuf = VITAAUD_GetDeviceBuf;
+    impl->CloseDevice = VITAAUD_CloseDevice;
+    impl->ThreadInit = VITAAUD_ThreadInit;
+
+    impl->CaptureFromDevice = VITAAUD_CaptureFromDevice;
+
+    /* and the capabilities */
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap VITAAUD_bootstrap = {
+    "vita", "VITA audio driver", VITAAUD_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_VITA */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/vita/SDL_vitaaudio.h

@@ -0,0 +1,46 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef _SDL_vitaaudio_h
+#define _SDL_vitaaudio_h
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS SDL_AudioDevice *this
+
+#define NUM_BUFFERS 2
+
+struct SDL_PrivateAudioData
+{
+    /* The hardware input/output port. */
+    int port;
+    /* The raw allocated mixing buffer. */
+    Uint8 *rawbuf;
+    /* Individual mixing buffers. */
+    Uint8 *mixbufs[NUM_BUFFERS];
+    /* Index of the next available mixing buffer. */
+    int next_buffer;
+};
+
+#endif /* _SDL_vitaaudio_h */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/wasapi/SDL_wasapi.c

@@ -0,0 +1,622 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_WASAPI
+
+#include "../../core/windows/SDL_windows.h"
+#include "../../core/windows/SDL_immdevice.h"
+#include "SDL_audio.h"
+#include "SDL_timer.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_sysaudio.h"
+
+#define COBJMACROS
+#include <audioclient.h>
+
+#include "SDL_wasapi.h"
+
+/* These constants aren't available in older SDKs */
+#ifndef AUDCLNT_STREAMFLAGS_RATEADJUST
+#define AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000
+#endif
+#ifndef AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY
+#define AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000
+#endif
+#ifndef AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM
+#define AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000
+#endif
+
+/* Some GUIDs we need to know without linking to libraries that aren't available before Vista. */
+static const IID SDL_IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483, { 0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2 } };
+static const IID SDL_IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0, { 0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17 } };
+
+
+static void WASAPI_DetectDevices(void)
+{
+    WASAPI_EnumerateEndpoints();
+}
+
+static SDL_INLINE SDL_bool WasapiFailed(_THIS, const HRESULT err)
+{
+    if (err == S_OK) {
+        return SDL_FALSE;
+    }
+
+    if (err == AUDCLNT_E_DEVICE_INVALIDATED) {
+        this->hidden->device_lost = SDL_TRUE;
+    } else if (SDL_AtomicGet(&this->enabled)) {
+        IAudioClient_Stop(this->hidden->client);
+        SDL_OpenedAudioDeviceDisconnected(this);
+        SDL_assert(!SDL_AtomicGet(&this->enabled));
+    }
+
+    return SDL_TRUE;
+}
+
+static int UpdateAudioStream(_THIS, const SDL_AudioSpec *oldspec)
+{
+    /* Since WASAPI requires us to handle all audio conversion, and our
+       device format might have changed, we might have to add/remove/change
+       the audio stream that the higher level uses to convert data, so
+       SDL keeps firing the callback as if nothing happened here. */
+
+    if ((this->callbackspec.channels == this->spec.channels) &&
+        (this->callbackspec.format == this->spec.format) &&
+        (this->callbackspec.freq == this->spec.freq) &&
+        (this->callbackspec.samples == this->spec.samples)) {
+        /* no need to buffer/convert in an AudioStream! */
+        SDL_FreeAudioStream(this->stream);
+        this->stream = NULL;
+    } else if ((oldspec->channels == this->spec.channels) &&
+               (oldspec->format == this->spec.format) &&
+               (oldspec->freq == this->spec.freq)) {
+        /* The existing audio stream is okay to keep using. */
+    } else {
+        /* replace the audiostream for new format */
+        SDL_FreeAudioStream(this->stream);
+        if (this->iscapture) {
+            this->stream = SDL_NewAudioStream(this->spec.format,
+                                              this->spec.channels, this->spec.freq,
+                                              this->callbackspec.format,
+                                              this->callbackspec.channels,
+                                              this->callbackspec.freq);
+        } else {
+            this->stream = SDL_NewAudioStream(this->callbackspec.format,
+                                              this->callbackspec.channels,
+                                              this->callbackspec.freq, this->spec.format,
+                                              this->spec.channels, this->spec.freq);
+        }
+
+        if (!this->stream) {
+            return -1; /* SDL_NewAudioStream should have called SDL_SetError. */
+        }
+    }
+
+    /* make sure our scratch buffer can cover the new device spec. */
+    if (this->spec.size > this->work_buffer_len) {
+        Uint8 *ptr = (Uint8 *)SDL_realloc(this->work_buffer, this->spec.size);
+        if (ptr == NULL) {
+            return SDL_OutOfMemory();
+        }
+        this->work_buffer = ptr;
+        this->work_buffer_len = this->spec.size;
+    }
+
+    return 0;
+}
+
+static void ReleaseWasapiDevice(_THIS);
+
+static SDL_bool RecoverWasapiDevice(_THIS)
+{
+    ReleaseWasapiDevice(this); /* dump the lost device's handles. */
+
+    if (this->hidden->default_device_generation) {
+        this->hidden->default_device_generation = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
+    }
+
+    /* this can fail for lots of reasons, but the most likely is we had a
+       non-default device that was disconnected, so we can't recover. Default
+       devices try to reinitialize whatever the new default is, so it's more
+       likely to carry on here, but this handles a non-default device that
+       simply had its format changed in the Windows Control Panel. */
+    if (WASAPI_ActivateDevice(this, SDL_TRUE) == -1) {
+        SDL_OpenedAudioDeviceDisconnected(this);
+        return SDL_FALSE;
+    }
+
+    this->hidden->device_lost = SDL_FALSE;
+
+    return SDL_TRUE; /* okay, carry on with new device details! */
+}
+
+static SDL_bool RecoverWasapiIfLost(_THIS)
+{
+    const int generation = this->hidden->default_device_generation;
+    SDL_bool lost = this->hidden->device_lost;
+
+    if (!SDL_AtomicGet(&this->enabled)) {
+        return SDL_FALSE; /* already failed. */
+    }
+
+    if (!this->hidden->client) {
+        return SDL_TRUE; /* still waiting for activation. */
+    }
+
+    if (!lost && (generation > 0)) { /* is a default device? */
+        const int newgen = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
+        if (generation != newgen) { /* the desired default device was changed, jump over to it. */
+            lost = SDL_TRUE;
+        }
+    }
+
+    return lost ? RecoverWasapiDevice(this) : SDL_TRUE;
+}
+
+static Uint8 *WASAPI_GetDeviceBuf(_THIS)
+{
+    /* get an endpoint buffer from WASAPI. */
+    BYTE *buffer = NULL;
+
+    while (RecoverWasapiIfLost(this) && this->hidden->render) {
+        if (!WasapiFailed(this, IAudioRenderClient_GetBuffer(this->hidden->render, this->spec.samples, &buffer))) {
+            return (Uint8 *)buffer;
+        }
+        SDL_assert(buffer == NULL);
+    }
+
+    return (Uint8 *)buffer;
+}
+
+static void WASAPI_PlayDevice(_THIS)
+{
+    if (this->hidden->render != NULL) { /* definitely activated? */
+        /* WasapiFailed() will mark the device for reacquisition or removal elsewhere. */
+        WasapiFailed(this, IAudioRenderClient_ReleaseBuffer(this->hidden->render, this->spec.samples, 0));
+    }
+}
+
+static void WASAPI_WaitDevice(_THIS)
+{
+    while (RecoverWasapiIfLost(this) && this->hidden->client && this->hidden->event) {
+        DWORD waitResult = WaitForSingleObjectEx(this->hidden->event, 200, FALSE);
+        if (waitResult == WAIT_OBJECT_0) {
+            const UINT32 maxpadding = this->spec.samples;
+            UINT32 padding = 0;
+            if (!WasapiFailed(this, IAudioClient_GetCurrentPadding(this->hidden->client, &padding))) {
+                /*SDL_Log("WASAPI EVENT! padding=%u maxpadding=%u", (unsigned int)padding, (unsigned int)maxpadding);*/
+                if (this->iscapture) {
+                    if (padding > 0) {
+                        break;
+                    }
+                } else {
+                    if (padding <= maxpadding) {
+                        break;
+                    }
+                }
+            }
+        } else if (waitResult != WAIT_TIMEOUT) {
+            /*SDL_Log("WASAPI FAILED EVENT!");*/
+            IAudioClient_Stop(this->hidden->client);
+            SDL_OpenedAudioDeviceDisconnected(this);
+        }
+    }
+}
+
+static int WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    SDL_AudioStream *stream = this->hidden->capturestream;
+    const int avail = SDL_AudioStreamAvailable(stream);
+    if (avail > 0) {
+        const int cpy = SDL_min(buflen, avail);
+        SDL_AudioStreamGet(stream, buffer, cpy);
+        return cpy;
+    }
+
+    while (RecoverWasapiIfLost(this)) {
+        HRESULT ret;
+        BYTE *ptr = NULL;
+        UINT32 frames = 0;
+        DWORD flags = 0;
+
+        /* uhoh, client isn't activated yet, just return silence. */
+        if (!this->hidden->capture) {
+            /* Delay so we run at about the speed that audio would be arriving. */
+            SDL_Delay(((this->spec.samples * 1000) / this->spec.freq));
+            SDL_memset(buffer, this->spec.silence, buflen);
+            return buflen;
+        }
+
+        ret = IAudioCaptureClient_GetBuffer(this->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
+        if (ret != AUDCLNT_S_BUFFER_EMPTY) {
+            WasapiFailed(this, ret); /* mark device lost/failed if necessary. */
+        }
+
+        if ((ret == AUDCLNT_S_BUFFER_EMPTY) || !frames) {
+            WASAPI_WaitDevice(this);
+        } else if (ret == S_OK) {
+            const int total = ((int)frames) * this->hidden->framesize;
+            const int cpy = SDL_min(buflen, total);
+            const int leftover = total - cpy;
+            const SDL_bool silent = (flags & AUDCLNT_BUFFERFLAGS_SILENT) ? SDL_TRUE : SDL_FALSE;
+
+            if (silent) {
+                SDL_memset(buffer, this->spec.silence, cpy);
+            } else {
+                SDL_memcpy(buffer, ptr, cpy);
+            }
+
+            if (leftover > 0) {
+                ptr += cpy;
+                if (silent) {
+                    SDL_memset(ptr, this->spec.silence, leftover); /* I guess this is safe? */
+                }
+
+                if (SDL_AudioStreamPut(stream, ptr, leftover) == -1) {
+                    return -1; /* uhoh, out of memory, etc. Kill device.  :( */
+                }
+            }
+
+            ret = IAudioCaptureClient_ReleaseBuffer(this->hidden->capture, frames);
+            WasapiFailed(this, ret); /* mark device lost/failed if necessary. */
+
+            return cpy;
+        }
+    }
+
+    return -1; /* unrecoverable error. */
+}
+
+static void WASAPI_FlushCapture(_THIS)
+{
+    BYTE *ptr = NULL;
+    UINT32 frames = 0;
+    DWORD flags = 0;
+
+    if (!this->hidden->capture) {
+        return; /* not activated yet? */
+    }
+
+    /* just read until we stop getting packets, throwing them away. */
+    while (SDL_TRUE) {
+        const HRESULT ret = IAudioCaptureClient_GetBuffer(this->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
+        if (ret == AUDCLNT_S_BUFFER_EMPTY) {
+            break; /* no more buffered data; we're done. */
+        } else if (WasapiFailed(this, ret)) {
+            break; /* failed for some other reason, abort. */
+        } else if (WasapiFailed(this, IAudioCaptureClient_ReleaseBuffer(this->hidden->capture, frames))) {
+            break; /* something broke. */
+        }
+    }
+    SDL_AudioStreamClear(this->hidden->capturestream);
+}
+
+static void ReleaseWasapiDevice(_THIS)
+{
+    if (this->hidden->client) {
+        IAudioClient_Stop(this->hidden->client);
+        IAudioClient_Release(this->hidden->client);
+        this->hidden->client = NULL;
+    }
+
+    if (this->hidden->render) {
+        IAudioRenderClient_Release(this->hidden->render);
+        this->hidden->render = NULL;
+    }
+
+    if (this->hidden->capture) {
+        IAudioCaptureClient_Release(this->hidden->capture);
+        this->hidden->capture = NULL;
+    }
+
+    if (this->hidden->waveformat) {
+        CoTaskMemFree(this->hidden->waveformat);
+        this->hidden->waveformat = NULL;
+    }
+
+    if (this->hidden->capturestream) {
+        SDL_FreeAudioStream(this->hidden->capturestream);
+        this->hidden->capturestream = NULL;
+    }
+
+    if (this->hidden->activation_handler) {
+        WASAPI_PlatformDeleteActivationHandler(this->hidden->activation_handler);
+        this->hidden->activation_handler = NULL;
+    }
+
+    if (this->hidden->event) {
+        CloseHandle(this->hidden->event);
+        this->hidden->event = NULL;
+    }
+}
+
+static void WASAPI_CloseDevice(_THIS)
+{
+    WASAPI_UnrefDevice(this);
+}
+
+void WASAPI_RefDevice(_THIS)
+{
+    SDL_AtomicIncRef(&this->hidden->refcount);
+}
+
+void WASAPI_UnrefDevice(_THIS)
+{
+    if (!SDL_AtomicDecRef(&this->hidden->refcount)) {
+        return;
+    }
+
+    /* actual closing happens here. */
+
+    /* don't touch this->hidden->task in here; it has to be reverted from
+       our callback thread. We do that in WASAPI_ThreadDeinit().
+       (likewise for this->hidden->coinitialized). */
+    ReleaseWasapiDevice(this);
+
+    if (SDL_ThreadID() == this->hidden->open_threadid) {
+        WIN_CoUninitialize();  /* if you closed from a different thread than you opened, sorry, it's a leak. We can't help you. */
+    }
+
+    SDL_free(this->hidden->devid);
+    SDL_free(this->hidden);
+}
+
+/* This is called once a device is activated, possibly asynchronously. */
+int WASAPI_PrepDevice(_THIS, const SDL_bool updatestream)
+{
+    /* !!! FIXME: we could request an exclusive mode stream, which is lower latency;
+       !!!  it will write into the kernel's audio buffer directly instead of
+       !!!  shared memory that a user-mode mixer then writes to the kernel with
+       !!!  everything else. Doing this means any other sound using this device will
+       !!!  stop playing, including the user's MP3 player and system notification
+       !!!  sounds. You'd probably need to release the device when the app isn't in
+       !!!  the foreground, to be a good citizen of the system. It's doable, but it's
+       !!!  more work and causes some annoyances, and I don't know what the latency
+       !!!  wins actually look like. Maybe add a hint to force exclusive mode at
+       !!!  some point. To be sure, defaulting to shared mode is the right thing to
+       !!!  do in any case. */
+    const SDL_AudioSpec oldspec = this->spec;
+    const AUDCLNT_SHAREMODE sharemode = AUDCLNT_SHAREMODE_SHARED;
+    UINT32 bufsize = 0; /* this is in sample frames, not samples, not bytes. */
+    REFERENCE_TIME default_period = 0;
+    IAudioClient *client = this->hidden->client;
+    IAudioRenderClient *render = NULL;
+    IAudioCaptureClient *capture = NULL;
+    WAVEFORMATEX *waveformat = NULL;
+    SDL_AudioFormat test_format;
+    SDL_AudioFormat wasapi_format = 0;
+    HRESULT ret = S_OK;
+    DWORD streamflags = 0;
+
+    SDL_assert(client != NULL);
+
+#if defined(__WINRT__) || defined(__GDK__) /* CreateEventEx() arrived in Vista, so we need an #ifdef for XP. */
+    this->hidden->event = CreateEventEx(NULL, NULL, 0, EVENT_ALL_ACCESS);
+#else
+    this->hidden->event = CreateEventW(NULL, 0, 0, NULL);
+#endif
+
+    if (this->hidden->event == NULL) {
+        return WIN_SetError("WASAPI can't create an event handle");
+    }
+
+    ret = IAudioClient_GetMixFormat(client, &waveformat);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't determine mix format", ret);
+    }
+
+    SDL_assert(waveformat != NULL);
+    this->hidden->waveformat = waveformat;
+
+    this->spec.channels = (Uint8)waveformat->nChannels;
+
+    /* Make sure we have a valid format that we can convert to whatever WASAPI wants. */
+    wasapi_format = WaveFormatToSDLFormat(waveformat);
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        if (test_format == wasapi_format) {
+            this->spec.format = test_format;
+            break;
+        }
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "wasapi");
+    }
+
+    ret = IAudioClient_GetDevicePeriod(client, &default_period, NULL);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't determine minimum device period", ret);
+    }
+
+    /* we've gotten reports that WASAPI's resampler introduces distortions, but in the short term
+       it fixes some other WASAPI-specific quirks we haven't quite tracked down.
+       Refer to bug #6326 for the immediate concern. */
+#if 0
+    this->spec.freq = waveformat->nSamplesPerSec;  /* force sampling rate so our resampler kicks in, if necessary. */
+#else
+    /* favor WASAPI's resampler over our own */
+    if (this->spec.freq != waveformat->nSamplesPerSec) {
+        streamflags |= (AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY);
+        waveformat->nSamplesPerSec = this->spec.freq;
+        waveformat->nAvgBytesPerSec = waveformat->nSamplesPerSec * waveformat->nChannels * (waveformat->wBitsPerSample / 8);
+    }
+#endif
+
+    streamflags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
+    ret = IAudioClient_Initialize(client, sharemode, streamflags, 0, 0, waveformat, NULL);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't initialize audio client", ret);
+    }
+
+    ret = IAudioClient_SetEventHandle(client, this->hidden->event);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't set event handle", ret);
+    }
+
+    ret = IAudioClient_GetBufferSize(client, &bufsize);
+    if (FAILED(ret)) {
+        return WIN_SetErrorFromHRESULT("WASAPI can't determine buffer size", ret);
+    }
+
+    /* Match the callback size to the period size to cut down on the number of
+       interrupts waited for in each call to WaitDevice */
+    {
+        const float period_millis = default_period / 10000.0f;
+        const float period_frames = period_millis * this->spec.freq / 1000.0f;
+        this->spec.samples = (Uint16)SDL_ceilf(period_frames);
+    }
+
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    this->hidden->framesize = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
+
+    if (this->iscapture) {
+        this->hidden->capturestream = SDL_NewAudioStream(this->spec.format, this->spec.channels, this->spec.freq, this->spec.format, this->spec.channels, this->spec.freq);
+        if (!this->hidden->capturestream) {
+            return -1; /* already set SDL_Error */
+        }
+
+        ret = IAudioClient_GetService(client, &SDL_IID_IAudioCaptureClient, (void **)&capture);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't get capture client service", ret);
+        }
+
+        SDL_assert(capture != NULL);
+        this->hidden->capture = capture;
+        ret = IAudioClient_Start(client);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't start capture", ret);
+        }
+
+        WASAPI_FlushCapture(this); /* MSDN says you should flush capture endpoint right after startup. */
+    } else {
+        ret = IAudioClient_GetService(client, &SDL_IID_IAudioRenderClient, (void **)&render);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't get render client service", ret);
+        }
+
+        SDL_assert(render != NULL);
+        this->hidden->render = render;
+        ret = IAudioClient_Start(client);
+        if (FAILED(ret)) {
+            return WIN_SetErrorFromHRESULT("WASAPI can't start playback", ret);
+        }
+    }
+
+    if (updatestream) {
+        return UpdateAudioStream(this, &oldspec);
+    }
+
+    return 0; /* good to go. */
+}
+
+static int WASAPI_OpenDevice(_THIS, const char *devname)
+{
+    LPCWSTR devid = (LPCWSTR)this->handle;
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *) SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    WASAPI_RefDevice(this); /* so CloseDevice() will unref to zero. */
+
+    if (FAILED(WIN_CoInitialize())) {  /* WASAPI uses COM, we need to make sure it's initialized. You have to close the device from the same thread!! */
+        return SDL_SetError("WIN_CoInitialize failed during WASAPI device open");
+    }
+    this->hidden->open_threadid = SDL_ThreadID();  /* set this _after_ coinitialize so we don't uninit if device fails at the wrong moment. */
+
+    if (!devid) { /* is default device? */
+        this->hidden->default_device_generation = SDL_AtomicGet(this->iscapture ? &SDL_IMMDevice_DefaultCaptureGeneration : &SDL_IMMDevice_DefaultPlaybackGeneration);
+    } else {
+        this->hidden->devid = SDL_wcsdup(devid);
+        if (!this->hidden->devid) {
+            return SDL_OutOfMemory();
+        }
+    }
+
+    if (WASAPI_ActivateDevice(this, SDL_FALSE) == -1) {
+        return -1; /* already set error. */
+    }
+
+    /* Ready, but waiting for async device activation.
+       Until activation is successful, we will report silence from capture
+       devices and ignore data on playback devices.
+       Also, since we don't know the _actual_ device format until after
+       activation, we let the app have whatever it asks for. We set up
+       an SDL_AudioStream to convert, if necessary, once the activation
+       completes. */
+
+    return 0;
+}
+
+static void WASAPI_ThreadInit(_THIS)
+{
+    WASAPI_PlatformThreadInit(this);
+}
+
+static void WASAPI_ThreadDeinit(_THIS)
+{
+    WASAPI_PlatformThreadDeinit(this);
+}
+
+static void WASAPI_Deinitialize(void)
+{
+    WASAPI_PlatformDeinit();
+}
+
+static SDL_bool WASAPI_Init(SDL_AudioDriverImpl *impl)
+{
+    if (WASAPI_PlatformInit() == -1) {
+        return SDL_FALSE;
+    }
+
+    /* Set the function pointers */
+    impl->DetectDevices = WASAPI_DetectDevices;
+    impl->ThreadInit = WASAPI_ThreadInit;
+    impl->ThreadDeinit = WASAPI_ThreadDeinit;
+    impl->OpenDevice = WASAPI_OpenDevice;
+    impl->PlayDevice = WASAPI_PlayDevice;
+    impl->WaitDevice = WASAPI_WaitDevice;
+    impl->GetDeviceBuf = WASAPI_GetDeviceBuf;
+    impl->CaptureFromDevice = WASAPI_CaptureFromDevice;
+    impl->FlushCapture = WASAPI_FlushCapture;
+    impl->CloseDevice = WASAPI_CloseDevice;
+    impl->Deinitialize = WASAPI_Deinitialize;
+    impl->GetDefaultAudioInfo = WASAPI_GetDefaultAudioInfo;
+    impl->HasCaptureSupport = SDL_TRUE;
+    impl->SupportsNonPow2Samples = SDL_TRUE;
+
+    return SDL_TRUE; /* this audio target is available. */
+}
+
+AudioBootStrap WASAPI_bootstrap = {
+    "wasapi", "WASAPI", WASAPI_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_WASAPI */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/wasapi/SDL_wasapi.h

@@ -0,0 +1,80 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_wasapi_h_
+#define SDL_wasapi_h_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#ifdef __cplusplus
+#define _THIS SDL_AudioDevice *_this
+#else
+#define _THIS SDL_AudioDevice *this
+#endif
+
+struct SDL_PrivateAudioData
+{
+    SDL_atomic_t refcount;
+    WCHAR *devid;
+    WAVEFORMATEX *waveformat;
+    IAudioClient *client;
+    IAudioRenderClient *render;
+    IAudioCaptureClient *capture;
+    SDL_AudioStream *capturestream;
+    HANDLE event;
+    HANDLE task;
+    SDL_threadID open_threadid;
+    SDL_bool coinitialized;
+    int framesize;
+    int default_device_generation;
+    SDL_bool device_lost;
+    void *activation_handler;
+    SDL_atomic_t just_activated;
+};
+
+/* win32 and winrt implementations call into these. */
+int WASAPI_PrepDevice(_THIS, const SDL_bool updatestream);
+void WASAPI_RefDevice(_THIS);
+void WASAPI_UnrefDevice(_THIS);
+
+/* These are functions that are implemented differently for Windows vs WinRT. */
+int WASAPI_PlatformInit(void);
+void WASAPI_PlatformDeinit(void);
+void WASAPI_EnumerateEndpoints(void);
+int WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture);
+int WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery);
+void WASAPI_PlatformThreadInit(_THIS);
+void WASAPI_PlatformThreadDeinit(_THIS);
+void WASAPI_PlatformDeleteActivationHandler(void *handler);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* SDL_wasapi_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/wasapi/SDL_wasapi_win32.c

@@ -0,0 +1,153 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+/* This is code that Windows uses to talk to WASAPI-related system APIs.
+   This is for non-WinRT desktop apps. The C++/CX implementation of these
+   functions, exclusive to WinRT, are in SDL_wasapi_winrt.cpp.
+   The code in SDL_wasapi.c is used by both standard Windows and WinRT builds
+   to deal with audio and calls into these functions. */
+
+#if SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__)
+
+#include "../../core/windows/SDL_windows.h"
+#include "../../core/windows/SDL_immdevice.h"
+#include "SDL_audio.h"
+#include "SDL_timer.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_sysaudio.h"
+
+#include <audioclient.h>
+
+#include "SDL_wasapi.h"
+
+/* handle to Avrt.dll--Vista and later!--for flagging the callback thread as "Pro Audio" (low latency). */
+static HMODULE libavrt = NULL;
+typedef HANDLE(WINAPI *pfnAvSetMmThreadCharacteristicsW)(LPCWSTR, LPDWORD);
+typedef BOOL(WINAPI *pfnAvRevertMmThreadCharacteristics)(HANDLE);
+static pfnAvSetMmThreadCharacteristicsW pAvSetMmThreadCharacteristicsW = NULL;
+static pfnAvRevertMmThreadCharacteristics pAvRevertMmThreadCharacteristics = NULL;
+
+/* Some GUIDs we need to know without linking to libraries that aren't available before Vista. */
+static const IID SDL_IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, { 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2 } };
+
+int WASAPI_PlatformInit(void)
+{
+    if (SDL_IMMDevice_Init() < 0) {
+        return -1; /* This is set by SDL_IMMDevice_Init */
+    }
+
+    libavrt = LoadLibrary(TEXT("avrt.dll")); /* this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now! */
+    if (libavrt) {
+        pAvSetMmThreadCharacteristicsW = (pfnAvSetMmThreadCharacteristicsW)GetProcAddress(libavrt, "AvSetMmThreadCharacteristicsW");
+        pAvRevertMmThreadCharacteristics = (pfnAvRevertMmThreadCharacteristics)GetProcAddress(libavrt, "AvRevertMmThreadCharacteristics");
+    }
+
+    return 0;
+}
+
+void WASAPI_PlatformDeinit(void)
+{
+    if (libavrt) {
+        FreeLibrary(libavrt);
+        libavrt = NULL;
+    }
+
+    pAvSetMmThreadCharacteristicsW = NULL;
+    pAvRevertMmThreadCharacteristics = NULL;
+
+    SDL_IMMDevice_Quit();
+}
+
+void WASAPI_PlatformThreadInit(_THIS)
+{
+    /* this thread uses COM. */
+    if (SUCCEEDED(WIN_CoInitialize())) { /* can't report errors, hope it worked! */
+        this->hidden->coinitialized = SDL_TRUE;
+    }
+
+    /* Set this thread to very high "Pro Audio" priority. */
+    if (pAvSetMmThreadCharacteristicsW) {
+        DWORD idx = 0;
+        this->hidden->task = pAvSetMmThreadCharacteristicsW(L"Pro Audio", &idx);
+    }
+}
+
+void WASAPI_PlatformThreadDeinit(_THIS)
+{
+    /* Set this thread back to normal priority. */
+    if (this->hidden->task && pAvRevertMmThreadCharacteristics) {
+        pAvRevertMmThreadCharacteristics(this->hidden->task);
+        this->hidden->task = NULL;
+    }
+
+    if (this->hidden->coinitialized) {
+        WIN_CoUninitialize();
+        this->hidden->coinitialized = SDL_FALSE;
+    }
+}
+
+int WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
+{
+    IMMDevice *device = NULL;
+    HRESULT ret;
+
+    if (SDL_IMMDevice_Get(this->hidden->devid, &device, this->iscapture) < 0) {
+        this->hidden->client = NULL;
+        return -1; /* This is already set by SDL_IMMDevice_Get */
+    }
+
+    /* this is not async in standard win32, yay! */
+    ret = IMMDevice_Activate(device, &SDL_IID_IAudioClient, CLSCTX_ALL, NULL, (void **)&this->hidden->client);
+    IMMDevice_Release(device);
+
+    if (FAILED(ret)) {
+        SDL_assert(this->hidden->client == NULL);
+        return WIN_SetErrorFromHRESULT("WASAPI can't activate audio endpoint", ret);
+    }
+
+    SDL_assert(this->hidden->client != NULL);
+    if (WASAPI_PrepDevice(this, isrecovery) == -1) { /* not async, fire it right away. */
+        return -1;
+    }
+
+    return 0; /* good to go. */
+}
+
+void WASAPI_EnumerateEndpoints(void)
+{
+    SDL_IMMDevice_EnumerateEndpoints(SDL_FALSE);
+}
+
+int WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
+{
+    return SDL_IMMDevice_GetDefaultAudioInfo(name, spec, iscapture);
+}
+
+void WASAPI_PlatformDeleteActivationHandler(void *handler)
+{
+    /* not asynchronous. */
+    SDL_assert(!"This function should have only been called on WinRT.");
+}
+
+#endif /* SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__) */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/wasapi/SDL_wasapi_winrt.cpp

@@ -0,0 +1,431 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+// This is C++/CX code that the WinRT port uses to talk to WASAPI-related
+//  system APIs. The C implementation of these functions, for non-WinRT apps,
+//  is in SDL_wasapi_win32.c. The code in SDL_wasapi.c is used by both standard
+//  Windows and WinRT builds to deal with audio and calls into these functions.
+
+#if SDL_AUDIO_DRIVER_WASAPI && defined(__WINRT__)
+
+#include <Windows.h>
+#include <windows.ui.core.h>
+#include <windows.devices.enumeration.h>
+#include <windows.media.devices.h>
+#include <wrl/implements.h>
+#include <collection.h>
+
+extern "C" {
+#include "../../core/windows/SDL_windows.h"
+#include "SDL_audio.h"
+#include "SDL_timer.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_sysaudio.h"
+}
+
+#define COBJMACROS
+#include <mmdeviceapi.h>
+#include <audioclient.h>
+
+#include "SDL_wasapi.h"
+
+using namespace Windows::Devices::Enumeration;
+using namespace Windows::Media::Devices;
+using namespace Windows::Foundation;
+using namespace Microsoft::WRL;
+
+static Platform::String ^ SDL_PKEY_AudioEngine_DeviceFormat = L"{f19f064d-082c-4e27-bc73-6882a1bb8e4c} 0";
+
+static void WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENSIBLE *fmt, LPCWSTR devid);
+static void WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid);
+extern "C" {
+SDL_atomic_t SDL_IMMDevice_DefaultPlaybackGeneration;
+SDL_atomic_t SDL_IMMDevice_DefaultCaptureGeneration;
+}
+
+/* This is a list of device id strings we have inflight, so we have consistent pointers to the same device. */
+typedef struct DevIdList
+{
+    WCHAR *str;
+    struct DevIdList *next;
+} DevIdList;
+
+static DevIdList *deviceid_list = NULL;
+
+class SDL_WasapiDeviceEventHandler
+{
+  public:
+    SDL_WasapiDeviceEventHandler(const SDL_bool _iscapture);
+    ~SDL_WasapiDeviceEventHandler();
+    void OnDeviceAdded(DeviceWatcher ^ sender, DeviceInformation ^ args);
+    void OnDeviceRemoved(DeviceWatcher ^ sender, DeviceInformationUpdate ^ args);
+    void OnDeviceUpdated(DeviceWatcher ^ sender, DeviceInformationUpdate ^ args);
+    void OnEnumerationCompleted(DeviceWatcher ^ sender, Platform::Object ^ args);
+    void OnDefaultRenderDeviceChanged(Platform::Object ^ sender, DefaultAudioRenderDeviceChangedEventArgs ^ args);
+    void OnDefaultCaptureDeviceChanged(Platform::Object ^ sender, DefaultAudioCaptureDeviceChangedEventArgs ^ args);
+    SDL_semaphore *completed;
+
+  private:
+    const SDL_bool iscapture;
+    DeviceWatcher ^ watcher;
+    Windows::Foundation::EventRegistrationToken added_handler;
+    Windows::Foundation::EventRegistrationToken removed_handler;
+    Windows::Foundation::EventRegistrationToken updated_handler;
+    Windows::Foundation::EventRegistrationToken completed_handler;
+    Windows::Foundation::EventRegistrationToken default_changed_handler;
+};
+
+SDL_WasapiDeviceEventHandler::SDL_WasapiDeviceEventHandler(const SDL_bool _iscapture)
+    : iscapture(_iscapture), completed(SDL_CreateSemaphore(0))
+{
+    if (!completed)
+        return; // uhoh.
+
+    Platform::String ^ selector = _iscapture ? MediaDevice::GetAudioCaptureSelector() : MediaDevice::GetAudioRenderSelector();
+    Platform::Collections::Vector<Platform::String ^> properties;
+    properties.Append(SDL_PKEY_AudioEngine_DeviceFormat);
+    watcher = DeviceInformation::CreateWatcher(selector, properties.GetView());
+    if (!watcher)
+        return; // uhoh.
+
+    // !!! FIXME: this doesn't need a lambda here, I think, if I make SDL_WasapiDeviceEventHandler a proper C++/CX class. --ryan.
+    added_handler = watcher->Added += ref new TypedEventHandler<DeviceWatcher ^, DeviceInformation ^>([this](DeviceWatcher ^ sender, DeviceInformation ^ args) { OnDeviceAdded(sender, args); });
+    removed_handler = watcher->Removed += ref new TypedEventHandler<DeviceWatcher ^, DeviceInformationUpdate ^>([this](DeviceWatcher ^ sender, DeviceInformationUpdate ^ args) { OnDeviceRemoved(sender, args); });
+    updated_handler = watcher->Updated += ref new TypedEventHandler<DeviceWatcher ^, DeviceInformationUpdate ^>([this](DeviceWatcher ^ sender, DeviceInformationUpdate ^ args) { OnDeviceUpdated(sender, args); });
+    completed_handler = watcher->EnumerationCompleted += ref new TypedEventHandler<DeviceWatcher ^, Platform::Object ^>([this](DeviceWatcher ^ sender, Platform::Object ^ args) { OnEnumerationCompleted(sender, args); });
+    if (iscapture) {
+        default_changed_handler = MediaDevice::DefaultAudioCaptureDeviceChanged += ref new TypedEventHandler<Platform::Object ^, DefaultAudioCaptureDeviceChangedEventArgs ^>([this](Platform::Object ^ sender, DefaultAudioCaptureDeviceChangedEventArgs ^ args) { OnDefaultCaptureDeviceChanged(sender, args); });
+    } else {
+        default_changed_handler = MediaDevice::DefaultAudioRenderDeviceChanged += ref new TypedEventHandler<Platform::Object ^, DefaultAudioRenderDeviceChangedEventArgs ^>([this](Platform::Object ^ sender, DefaultAudioRenderDeviceChangedEventArgs ^ args) { OnDefaultRenderDeviceChanged(sender, args); });
+    }
+    watcher->Start();
+}
+
+SDL_WasapiDeviceEventHandler::~SDL_WasapiDeviceEventHandler()
+{
+    if (watcher) {
+        watcher->Added -= added_handler;
+        watcher->Removed -= removed_handler;
+        watcher->Updated -= updated_handler;
+        watcher->EnumerationCompleted -= completed_handler;
+        watcher->Stop();
+        watcher = nullptr;
+    }
+    if (completed) {
+        SDL_DestroySemaphore(completed);
+        completed = nullptr;
+    }
+
+    if (iscapture) {
+        MediaDevice::DefaultAudioCaptureDeviceChanged -= default_changed_handler;
+    } else {
+        MediaDevice::DefaultAudioRenderDeviceChanged -= default_changed_handler;
+    }
+}
+
+void SDL_WasapiDeviceEventHandler::OnDeviceAdded(DeviceWatcher ^ sender, DeviceInformation ^ info)
+{
+    SDL_assert(sender == this->watcher);
+    char *utf8dev = WIN_StringToUTF8(info->Name->Data());
+    if (utf8dev) {
+        WAVEFORMATEXTENSIBLE fmt;
+        Platform::Object ^ obj = info->Properties->Lookup(SDL_PKEY_AudioEngine_DeviceFormat);
+        if (obj) {
+            IPropertyValue ^ property = (IPropertyValue ^) obj;
+            Platform::Array<unsigned char> ^ data;
+            property->GetUInt8Array(&data);
+            SDL_memcpy(&fmt, data->Data, SDL_min(data->Length, sizeof(WAVEFORMATEXTENSIBLE)));
+        } else {
+            SDL_zero(fmt);
+        }
+
+        WASAPI_AddDevice(this->iscapture, utf8dev, &fmt, info->Id->Data());
+        SDL_free(utf8dev);
+    }
+}
+
+void SDL_WasapiDeviceEventHandler::OnDeviceRemoved(DeviceWatcher ^ sender, DeviceInformationUpdate ^ info)
+{
+    SDL_assert(sender == this->watcher);
+    WASAPI_RemoveDevice(this->iscapture, info->Id->Data());
+}
+
+void SDL_WasapiDeviceEventHandler::OnDeviceUpdated(DeviceWatcher ^ sender, DeviceInformationUpdate ^ args)
+{
+    SDL_assert(sender == this->watcher);
+}
+
+void SDL_WasapiDeviceEventHandler::OnEnumerationCompleted(DeviceWatcher ^ sender, Platform::Object ^ args)
+{
+    SDL_assert(sender == this->watcher);
+    SDL_SemPost(this->completed);
+}
+
+void SDL_WasapiDeviceEventHandler::OnDefaultRenderDeviceChanged(Platform::Object ^ sender, DefaultAudioRenderDeviceChangedEventArgs ^ args)
+{
+    SDL_assert(!this->iscapture);
+    SDL_AtomicAdd(&SDL_IMMDevice_DefaultPlaybackGeneration, 1);
+}
+
+void SDL_WasapiDeviceEventHandler::OnDefaultCaptureDeviceChanged(Platform::Object ^ sender, DefaultAudioCaptureDeviceChangedEventArgs ^ args)
+{
+    SDL_assert(this->iscapture);
+    SDL_AtomicAdd(&SDL_IMMDevice_DefaultCaptureGeneration, 1);
+}
+
+static SDL_WasapiDeviceEventHandler *playback_device_event_handler;
+static SDL_WasapiDeviceEventHandler *capture_device_event_handler;
+
+int WASAPI_PlatformInit(void)
+{
+    SDL_AtomicSet(&SDL_IMMDevice_DefaultPlaybackGeneration, 1);
+    SDL_AtomicSet(&SDL_IMMDevice_DefaultCaptureGeneration, 1);
+    return 0;
+}
+
+void WASAPI_PlatformDeinit(void)
+{
+    DevIdList *devidlist;
+    DevIdList *next;
+
+    delete playback_device_event_handler;
+    playback_device_event_handler = nullptr;
+    delete capture_device_event_handler;
+    capture_device_event_handler = nullptr;
+
+    for (devidlist = deviceid_list; devidlist; devidlist = next) {
+        next = devidlist->next;
+        SDL_free(devidlist->str);
+        SDL_free(devidlist);
+    }
+    deviceid_list = NULL;
+}
+
+void WASAPI_EnumerateEndpoints(void)
+{
+    // DeviceWatchers will fire an Added event for each existing device at
+    //  startup, so we don't need to enumerate them separately before
+    //  listening for updates.
+    playback_device_event_handler = new SDL_WasapiDeviceEventHandler(SDL_FALSE);
+    capture_device_event_handler = new SDL_WasapiDeviceEventHandler(SDL_TRUE);
+    SDL_SemWait(playback_device_event_handler->completed);
+    SDL_SemWait(capture_device_event_handler->completed);
+}
+
+struct SDL_WasapiActivationHandler : public RuntimeClass<RuntimeClassFlags<ClassicCom>, FtmBase, IActivateAudioInterfaceCompletionHandler>
+{
+    SDL_WasapiActivationHandler() : device(nullptr) {}
+    STDMETHOD(ActivateCompleted)
+    (IActivateAudioInterfaceAsyncOperation *operation);
+    SDL_AudioDevice *device;
+};
+
+HRESULT
+SDL_WasapiActivationHandler::ActivateCompleted(IActivateAudioInterfaceAsyncOperation *async)
+{
+    // Just set a flag, since we're probably in a different thread. We'll pick it up and init everything on our own thread to prevent races.
+    SDL_AtomicSet(&device->hidden->just_activated, 1);
+    WASAPI_UnrefDevice(device);
+    return S_OK;
+}
+
+void WASAPI_PlatformDeleteActivationHandler(void *handler)
+{
+    ((SDL_WasapiActivationHandler *)handler)->Release();
+}
+
+int WASAPI_GetDefaultAudioInfo(char **name, SDL_AudioSpec *spec, int iscapture)
+{
+    return SDL_Unsupported();
+}
+
+int WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
+{
+    LPCWSTR devid = _this->hidden->devid;
+    Platform::String ^ defdevid;
+
+    if (devid == nullptr) {
+        defdevid = _this->iscapture ? MediaDevice::GetDefaultAudioCaptureId(AudioDeviceRole::Default) : MediaDevice::GetDefaultAudioRenderId(AudioDeviceRole::Default);
+        if (defdevid) {
+            devid = defdevid->Data();
+        }
+    }
+
+    SDL_AtomicSet(&_this->hidden->just_activated, 0);
+
+    ComPtr<SDL_WasapiActivationHandler> handler = Make<SDL_WasapiActivationHandler>();
+    if (handler == nullptr) {
+        return SDL_SetError("Failed to allocate WASAPI activation handler");
+    }
+
+    handler.Get()->AddRef(); // we hold a reference after ComPtr destructs on return, causing a Release, and Release ourselves in WASAPI_PlatformDeleteActivationHandler(), etc.
+    handler.Get()->device = _this;
+    _this->hidden->activation_handler = handler.Get();
+
+    WASAPI_RefDevice(_this); /* completion handler will unref it. */
+    IActivateAudioInterfaceAsyncOperation *async = nullptr;
+    const HRESULT ret = ActivateAudioInterfaceAsync(devid, __uuidof(IAudioClient), nullptr, handler.Get(), &async);
+
+    if (FAILED(ret) || async == nullptr) {
+        if (async != nullptr) {
+            async->Release();
+        }
+        handler.Get()->Release();
+        WASAPI_UnrefDevice(_this);
+        return WIN_SetErrorFromHRESULT("WASAPI can't activate requested audio endpoint", ret);
+    }
+
+    /* Spin until the async operation is complete.
+     * If we don't PrepDevice before leaving this function, the bug list gets LONG:
+     * - device.spec is not filled with the correct information
+     * - The 'obtained' spec will be wrong for ALLOW_CHANGE properties
+     * - SDL_AudioStreams will/will not be allocated at the right time
+     * - SDL_assert(device->callbackspec.size == device->spec.size) will fail
+     * - When the assert is ignored, skipping or a buffer overflow will occur
+     */
+    while (!SDL_AtomicCAS(&_this->hidden->just_activated, 1, 0)) {
+        SDL_Delay(1);
+    }
+
+    HRESULT activateRes = S_OK;
+    IUnknown *iunknown = nullptr;
+    const HRESULT getActivateRes = async->GetActivateResult(&activateRes, &iunknown);
+    async->Release();
+    if (FAILED(getActivateRes)) {
+        return WIN_SetErrorFromHRESULT("Failed to get WASAPI activate result", getActivateRes);
+    } else if (FAILED(activateRes)) {
+        return WIN_SetErrorFromHRESULT("Failed to activate WASAPI device", activateRes);
+    }
+
+    iunknown->QueryInterface(IID_PPV_ARGS(&_this->hidden->client));
+    if (!_this->hidden->client) {
+        return SDL_SetError("Failed to query WASAPI client interface");
+    }
+
+    if (WASAPI_PrepDevice(_this, isrecovery) == -1) {
+        return -1;
+    }
+
+    return 0;
+}
+
+void WASAPI_PlatformThreadInit(_THIS)
+{
+    // !!! FIXME: set this thread to "Pro Audio" priority.
+}
+
+void WASAPI_PlatformThreadDeinit(_THIS)
+{
+    // !!! FIXME: set this thread to "Pro Audio" priority.
+}
+
+/* Everything below was copied from SDL_wasapi.c, before it got moved to SDL_immdevice.c! */
+
+static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
+static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
+
+extern "C" SDL_AudioFormat
+WaveFormatToSDLFormat(WAVEFORMATEX *waveformat)
+{
+    if ((waveformat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) && (waveformat->wBitsPerSample == 32)) {
+        return AUDIO_F32SYS;
+    } else if ((waveformat->wFormatTag == WAVE_FORMAT_PCM) && (waveformat->wBitsPerSample == 16)) {
+        return AUDIO_S16SYS;
+    } else if ((waveformat->wFormatTag == WAVE_FORMAT_PCM) && (waveformat->wBitsPerSample == 32)) {
+        return AUDIO_S32SYS;
+    } else if (waveformat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
+        const WAVEFORMATEXTENSIBLE *ext = (const WAVEFORMATEXTENSIBLE *)waveformat;
+        if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, sizeof(GUID)) == 0) && (waveformat->wBitsPerSample == 32)) {
+            return AUDIO_F32SYS;
+        } else if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID)) == 0) && (waveformat->wBitsPerSample == 16)) {
+            return AUDIO_S16SYS;
+        } else if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID)) == 0) && (waveformat->wBitsPerSample == 32)) {
+            return AUDIO_S32SYS;
+        }
+    }
+    return 0;
+}
+
+static void WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid)
+{
+    DevIdList *i;
+    DevIdList *next;
+    DevIdList *prev = NULL;
+    for (i = deviceid_list; i; i = next) {
+        next = i->next;
+        if (SDL_wcscmp(i->str, devid) == 0) {
+            if (prev) {
+                prev->next = next;
+            } else {
+                deviceid_list = next;
+            }
+            SDL_RemoveAudioDevice(iscapture, i->str);
+            SDL_free(i->str);
+            SDL_free(i);
+        } else {
+            prev = i;
+        }
+    }
+}
+
+static void WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENSIBLE *fmt, LPCWSTR devid)
+{
+    DevIdList *devidlist;
+    SDL_AudioSpec spec;
+
+    /* You can have multiple endpoints on a device that are mutually exclusive ("Speakers" vs "Line Out" or whatever).
+       In a perfect world, things that are unplugged won't be in this collection. The only gotcha is probably for
+       phones and tablets, where you might have an internal speaker and a headphone jack and expect both to be
+       available and switch automatically. (!!! FIXME...?) */
+
+    /* see if we already have this one. */
+    for (devidlist = deviceid_list; devidlist; devidlist = devidlist->next) {
+        if (SDL_wcscmp(devidlist->str, devid) == 0) {
+            return; /* we already have this. */
+        }
+    }
+
+    devidlist = (DevIdList *)SDL_malloc(sizeof(*devidlist));
+    if (devidlist == NULL) {
+        return; /* oh well. */
+    }
+
+    devid = SDL_wcsdup(devid);
+    if (!devid) {
+        SDL_free(devidlist);
+        return; /* oh well. */
+    }
+
+    devidlist->str = (WCHAR *)devid;
+    devidlist->next = deviceid_list;
+    deviceid_list = devidlist;
+
+    SDL_zero(spec);
+    spec.channels = (Uint8)fmt->Format.nChannels;
+    spec.freq = fmt->Format.nSamplesPerSec;
+    spec.format = WaveFormatToSDLFormat((WAVEFORMATEX *)fmt);
+    SDL_AddAudioDevice(iscapture, devname, &spec, (void *)devid);
+}
+
+#endif // SDL_AUDIO_DRIVER_WASAPI && defined(__WINRT__)
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/winmm/SDL_winmm.c

@@ -0,0 +1,443 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#if SDL_AUDIO_DRIVER_WINMM
+
+/* Allow access to a raw mixing buffer */
+
+#include "../../core/windows/SDL_windows.h"
+#include <mmsystem.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_winmm.h"
+
+/* MinGW32 mmsystem.h doesn't include these structures */
+#if defined(__MINGW32__) && defined(_MMSYSTEM_H)
+
+typedef struct tagWAVEINCAPS2W
+{
+    WORD wMid;
+    WORD wPid;
+    MMVERSION vDriverVersion;
+    WCHAR szPname[MAXPNAMELEN];
+    DWORD dwFormats;
+    WORD wChannels;
+    WORD wReserved1;
+    GUID ManufacturerGuid;
+    GUID ProductGuid;
+    GUID NameGuid;
+} WAVEINCAPS2W,*PWAVEINCAPS2W,*NPWAVEINCAPS2W,*LPWAVEINCAPS2W;
+
+typedef struct tagWAVEOUTCAPS2W
+{
+    WORD wMid;
+    WORD wPid;
+    MMVERSION vDriverVersion;
+    WCHAR szPname[MAXPNAMELEN];
+    DWORD dwFormats;
+    WORD wChannels;
+    WORD wReserved1;
+    DWORD dwSupport;
+    GUID ManufacturerGuid;
+    GUID ProductGuid;
+    GUID NameGuid;
+} WAVEOUTCAPS2W,*PWAVEOUTCAPS2W,*NPWAVEOUTCAPS2W,*LPWAVEOUTCAPS2W;
+
+#endif /* defined(__MINGW32__) && defined(_MMSYSTEM_H) */
+
+#ifndef WAVE_FORMAT_IEEE_FLOAT
+#define WAVE_FORMAT_IEEE_FLOAT 0x0003
+#endif
+
+#define DETECT_DEV_IMPL(iscap, typ, capstyp) \
+static void DetectWave##typ##Devs(void) { \
+    const UINT iscapture = iscap ? 1 : 0; \
+    const UINT devcount = wave##typ##GetNumDevs(); \
+    capstyp##2W caps; \
+    SDL_AudioSpec spec; \
+    UINT i; \
+    SDL_zero(spec); \
+    for (i = 0; i < devcount; i++) { \
+        if (wave##typ##GetDevCaps(i,(LP##capstyp##W)&caps,sizeof(caps))==MMSYSERR_NOERROR) { \
+            char *name = WIN_LookupAudioDeviceName(caps.szPname,&caps.NameGuid); \
+            if (name != NULL) { \
+                /* Note that freq/format are not filled in, as this information \
+                 * is not provided by the caps struct! At best, we get possible \
+                 * sample formats, but not an _active_ format. \
+                 */ \
+                spec.channels = (Uint8)caps.wChannels; \
+                SDL_AddAudioDevice((int) iscapture, name, &spec, (void *) ((size_t) i+1)); \
+                SDL_free(name); \
+            } \
+        } \
+    } \
+}
+
+DETECT_DEV_IMPL(SDL_FALSE, Out, WAVEOUTCAPS)
+DETECT_DEV_IMPL(SDL_TRUE, In, WAVEINCAPS)
+
+static void WINMM_DetectDevices(void)
+{
+    DetectWaveInDevs();
+    DetectWaveOutDevs();
+}
+
+static void CALLBACK CaptureSound(HWAVEIN hwi, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2)
+{
+    SDL_AudioDevice *this = (SDL_AudioDevice *) dwInstance;
+
+    /* Only service "buffer is filled" messages */
+    if (uMsg != WIM_DATA)
+        return;
+
+    /* Signal that we have a new buffer of data */
+    ReleaseSemaphore(this->hidden->audio_sem, 1, NULL);
+}
+
+
+/* The Win32 callback for filling the WAVE device */
+static void CALLBACK FillSound(HWAVEOUT hwo, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2)
+{
+    SDL_AudioDevice *this = (SDL_AudioDevice *) dwInstance;
+
+    /* Only service "buffer done playing" messages */
+    if (uMsg != WOM_DONE)
+        return;
+
+    /* Signal that we are done playing a buffer */
+    ReleaseSemaphore(this->hidden->audio_sem, 1, NULL);
+}
+
+static int SetMMerror(const char *function, MMRESULT code)
+{
+    int len;
+    char errbuf[MAXERRORLENGTH];
+    wchar_t werrbuf[MAXERRORLENGTH];
+
+    SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: ", function);
+    len = SDL_static_cast(int, SDL_strlen(errbuf));
+
+    waveOutGetErrorText(code, werrbuf, MAXERRORLENGTH - len);
+    WideCharToMultiByte(CP_ACP, 0, werrbuf, -1, errbuf + len,
+                        MAXERRORLENGTH - len, NULL, NULL);
+
+    return SDL_SetError("%s", errbuf);
+}
+
+static void WINMM_WaitDevice(_THIS)
+{
+    /* Wait for an audio chunk to finish */
+    WaitForSingleObject(this->hidden->audio_sem, INFINITE);
+}
+
+static Uint8 *WINMM_GetDeviceBuf(_THIS)
+{
+    return (Uint8 *) (this->hidden->
+                      wavebuf[this->hidden->next_buffer].lpData);
+}
+
+static void WINMM_PlayDevice(_THIS)
+{
+    /* Queue it up */
+    waveOutWrite(this->hidden->hout,
+                 &this->hidden->wavebuf[this->hidden->next_buffer],
+                 sizeof(this->hidden->wavebuf[0]));
+    this->hidden->next_buffer = (this->hidden->next_buffer + 1) % NUM_BUFFERS;
+}
+
+static int WINMM_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    const int nextbuf = this->hidden->next_buffer;
+    MMRESULT result;
+
+    SDL_assert(buflen == this->spec.size);
+
+    /* Wait for an audio chunk to finish */
+    WaitForSingleObject(this->hidden->audio_sem, INFINITE);
+
+    /* Copy it to caller's buffer... */
+    SDL_memcpy(buffer, this->hidden->wavebuf[nextbuf].lpData, this->spec.size);
+
+    /* requeue the buffer that just finished. */
+    result = waveInAddBuffer(this->hidden->hin,
+                             &this->hidden->wavebuf[nextbuf],
+                             sizeof(this->hidden->wavebuf[nextbuf]));
+    if (result != MMSYSERR_NOERROR) {
+        return -1;  /* uhoh! Disable the device. */
+    }
+
+    /* queue the next buffer in sequence, next time. */
+    this->hidden->next_buffer = (nextbuf + 1) % NUM_BUFFERS;
+    return this->spec.size;
+}
+
+static void WINMM_FlushCapture(_THIS)
+{
+    /* Wait for an audio chunk to finish */
+    if (WaitForSingleObject(this->hidden->audio_sem, 0) == WAIT_OBJECT_0) {
+        const int nextbuf = this->hidden->next_buffer;
+        /* requeue the buffer that just finished without reading from it. */
+        waveInAddBuffer(this->hidden->hin,
+                        &this->hidden->wavebuf[nextbuf],
+                        sizeof(this->hidden->wavebuf[nextbuf]));
+        this->hidden->next_buffer = (nextbuf + 1) % NUM_BUFFERS;
+    }
+}
+
+static void WINMM_CloseDevice(_THIS)
+{
+    int i;
+
+    if (this->hidden->hout) {
+        waveOutReset(this->hidden->hout);
+
+        /* Clean up mixing buffers */
+        for (i = 0; i < NUM_BUFFERS; ++i) {
+            if (this->hidden->wavebuf[i].dwUser != 0xFFFF) {
+                waveOutUnprepareHeader(this->hidden->hout,
+                                       &this->hidden->wavebuf[i],
+                                       sizeof(this->hidden->wavebuf[i]));
+            }
+        }
+
+        waveOutClose(this->hidden->hout);
+    }
+
+    if (this->hidden->hin) {
+        waveInReset(this->hidden->hin);
+
+        /* Clean up mixing buffers */
+        for (i = 0; i < NUM_BUFFERS; ++i) {
+            if (this->hidden->wavebuf[i].dwUser != 0xFFFF) {
+                waveInUnprepareHeader(this->hidden->hin,
+                                       &this->hidden->wavebuf[i],
+                                       sizeof(this->hidden->wavebuf[i]));
+            }
+        }
+        waveInClose(this->hidden->hin);
+    }
+
+    if (this->hidden->audio_sem) {
+        CloseHandle(this->hidden->audio_sem);
+    }
+
+    SDL_free(this->hidden->mixbuf);
+    SDL_free(this->hidden);
+}
+
+static SDL_bool PrepWaveFormat(_THIS, UINT devId, WAVEFORMATEX *pfmt, const int iscapture)
+{
+    SDL_zerop(pfmt);
+
+    if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
+        pfmt->wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
+    } else {
+        pfmt->wFormatTag = WAVE_FORMAT_PCM;
+    }
+    pfmt->wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
+
+    pfmt->nChannels = this->spec.channels;
+    pfmt->nSamplesPerSec = this->spec.freq;
+    pfmt->nBlockAlign = pfmt->nChannels * (pfmt->wBitsPerSample / 8);
+    pfmt->nAvgBytesPerSec = pfmt->nSamplesPerSec * pfmt->nBlockAlign;
+
+    if (iscapture) {
+        return (waveInOpen(0, devId, pfmt, 0, 0, WAVE_FORMAT_QUERY) == 0);
+    } else {
+        return (waveOutOpen(0, devId, pfmt, 0, 0, WAVE_FORMAT_QUERY) == 0);
+    }
+}
+
+static int WINMM_OpenDevice(_THIS, const char *devname)
+{
+    SDL_AudioFormat test_format;
+    SDL_bool iscapture = this->iscapture;
+    void *handle = this->handle;
+    MMRESULT result;
+    WAVEFORMATEX waveformat;
+    UINT devId = WAVE_MAPPER;  /* WAVE_MAPPER == choose system's default */
+    UINT i;
+
+    if (handle != NULL) {  /* specific device requested? */
+        /* -1 because we increment the original value to avoid NULL. */
+        const size_t val = ((size_t) handle) - 1;
+        devId = (UINT) val;
+    }
+
+    /* Initialize all variables that we clean on shutdown */
+    this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(*this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(this->hidden);
+
+    /* Initialize the wavebuf structures for closing */
+    for (i = 0; i < NUM_BUFFERS; ++i)
+        this->hidden->wavebuf[i].dwUser = 0xFFFF;
+
+    if (this->spec.channels > 2)
+        this->spec.channels = 2;        /* !!! FIXME: is this right? */
+
+    for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
+        switch (test_format) {
+        case AUDIO_U8:
+        case AUDIO_S16:
+        case AUDIO_S32:
+        case AUDIO_F32:
+            this->spec.format = test_format;
+            if (PrepWaveFormat(this, devId, &waveformat, iscapture)) {
+                break;
+            }
+            continue;
+        default:
+            continue;
+        }
+        break;
+    }
+
+    if (!test_format) {
+        return SDL_SetError("%s: Unsupported audio format", "winmm");
+    }
+
+    /* Update the fragment size as size in bytes */
+    SDL_CalculateAudioSpec(&this->spec);
+
+    /* Open the audio device */
+    if (iscapture) {
+        result = waveInOpen(&this->hidden->hin, devId, &waveformat,
+                             (DWORD_PTR) CaptureSound, (DWORD_PTR) this,
+                             CALLBACK_FUNCTION);
+        if (result != MMSYSERR_NOERROR) {
+            return SetMMerror("waveInOpen()", result);
+        }
+    } else {
+        result = waveOutOpen(&this->hidden->hout, devId, &waveformat,
+                             (DWORD_PTR) FillSound, (DWORD_PTR) this,
+                             CALLBACK_FUNCTION);
+        if (result != MMSYSERR_NOERROR) {
+            return SetMMerror("waveOutOpen()", result);
+        }
+    }
+
+#ifdef SOUND_DEBUG
+    /* Check the sound device we retrieved */
+    {
+        if (iscapture) {
+            WAVEINCAPS caps;
+            result = waveInGetDevCaps((UINT) this->hidden->hout,
+                                      &caps, sizeof(caps));
+            if (result != MMSYSERR_NOERROR) {
+                return SetMMerror("waveInGetDevCaps()", result);
+            }
+            printf("Audio device: %s\n", caps.szPname);
+        } else {
+            WAVEOUTCAPS caps;
+            result = waveOutGetDevCaps((UINT) this->hidden->hout,
+                                       &caps, sizeof(caps));
+            if (result != MMSYSERR_NOERROR) {
+                return SetMMerror("waveOutGetDevCaps()", result);
+            }
+            printf("Audio device: %s\n", caps.szPname);
+        }
+    }
+#endif
+
+    /* Create the audio buffer semaphore */
+    this->hidden->audio_sem = CreateSemaphore(NULL, iscapture ? 0 : NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
+    if (this->hidden->audio_sem == NULL) {
+        return SDL_SetError("Couldn't create semaphore");
+    }
+
+    /* Create the sound buffers */
+    this->hidden->mixbuf =
+        (Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
+    if (this->hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    SDL_zeroa(this->hidden->wavebuf);
+    for (i = 0; i < NUM_BUFFERS; ++i) {
+        this->hidden->wavebuf[i].dwBufferLength = this->spec.size;
+        this->hidden->wavebuf[i].dwFlags = WHDR_DONE;
+        this->hidden->wavebuf[i].lpData =
+            (LPSTR) & this->hidden->mixbuf[i * this->spec.size];
+
+        if (iscapture) {
+            result = waveInPrepareHeader(this->hidden->hin,
+                                          &this->hidden->wavebuf[i],
+                                          sizeof(this->hidden->wavebuf[i]));
+            if (result != MMSYSERR_NOERROR) {
+                return SetMMerror("waveInPrepareHeader()", result);
+            }
+
+            result = waveInAddBuffer(this->hidden->hin,
+                                     &this->hidden->wavebuf[i],
+                                     sizeof(this->hidden->wavebuf[i]));
+            if (result != MMSYSERR_NOERROR) {
+                return SetMMerror("waveInAddBuffer()", result);
+            }
+        } else {
+            result = waveOutPrepareHeader(this->hidden->hout,
+                                          &this->hidden->wavebuf[i],
+                                          sizeof(this->hidden->wavebuf[i]));
+            if (result != MMSYSERR_NOERROR) {
+                return SetMMerror("waveOutPrepareHeader()", result);
+            }
+        }
+    }
+
+    if (iscapture) {
+        result = waveInStart(this->hidden->hin);
+        if (result != MMSYSERR_NOERROR) {
+            return SetMMerror("waveInStart()", result);
+        }
+    }
+
+    return 0;                   /* Ready to go! */
+}
+
+static SDL_bool WINMM_Init(SDL_AudioDriverImpl * impl)
+{
+    /* Set the function pointers */
+    impl->DetectDevices = WINMM_DetectDevices;
+    impl->OpenDevice = WINMM_OpenDevice;
+    impl->PlayDevice = WINMM_PlayDevice;
+    impl->WaitDevice = WINMM_WaitDevice;
+    impl->GetDeviceBuf = WINMM_GetDeviceBuf;
+    impl->CaptureFromDevice = WINMM_CaptureFromDevice;
+    impl->FlushCapture = WINMM_FlushCapture;
+    impl->CloseDevice = WINMM_CloseDevice;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    return SDL_TRUE;   /* this audio target is available. */
+}
+
+AudioBootStrap WINMM_bootstrap = {
+    "winmm", "Windows Waveform Audio", WINMM_Init, SDL_FALSE
+};
+
+#endif /* SDL_AUDIO_DRIVER_WINMM */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/audio/winmm/SDL_winmm.h

@@ -0,0 +1,45 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+
+#ifndef SDL_winmm_h_
+#define SDL_winmm_h_
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the audio functions */
+#define _THIS   SDL_AudioDevice *this
+
+#define NUM_BUFFERS 2           /* -- Don't lower this! */
+
+struct SDL_PrivateAudioData
+{
+    HWAVEOUT hout;
+    HWAVEIN hin;
+    HANDLE audio_sem;
+    Uint8 *mixbuf;              /* The raw allocated mixing buffer */
+    WAVEHDR wavebuf[NUM_BUFFERS];       /* Wave audio fragments */
+    int next_buffer;
+};
+
+#endif /* SDL_winmm_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/core/android/SDL_android.h

@@ -0,0 +1,156 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../../SDL_internal.h"
+#include "SDL_system.h"
+
+/* Set up for C function definitions, even when using C++ */
+#ifdef __cplusplus
+/* *INDENT-OFF* */
+extern "C" {
+/* *INDENT-ON* */
+#endif
+
+#include <EGL/eglplatform.h>
+#include <android/native_window_jni.h>
+
+#include "SDL_audio.h"
+#include "SDL_rect.h"
+#include "SDL_video.h"
+
+/* Interface from the SDL library into the Android Java activity */
+extern void Android_JNI_SetActivityTitle(const char *title);
+extern void Android_JNI_SetWindowStyle(SDL_bool fullscreen);
+extern void Android_JNI_SetOrientation(int w, int h, int resizable, const char *hint);
+extern void Android_JNI_MinizeWindow(void);
+extern SDL_bool Android_JNI_ShouldMinimizeOnFocusLoss(void);
+
+extern SDL_bool Android_JNI_GetAccelerometerValues(float values[3]);
+extern void Android_JNI_ShowTextInput(SDL_Rect *inputRect);
+extern void Android_JNI_HideTextInput(void);
+extern SDL_bool Android_JNI_IsScreenKeyboardShown(void);
+extern ANativeWindow *Android_JNI_GetNativeWindow(void);
+
+extern SDL_DisplayOrientation Android_JNI_GetDisplayOrientation(void);
+extern int Android_JNI_GetDisplayDPI(float *ddpi, float *xdpi, float *ydpi);
+
+/* Audio support */
+extern void Android_DetectDevices(void);
+extern int Android_JNI_OpenAudioDevice(int iscapture, int device_id, SDL_AudioSpec *spec);
+extern void *Android_JNI_GetAudioBuffer(void);
+extern void Android_JNI_WriteAudioBuffer(void);
+extern int Android_JNI_CaptureAudioBuffer(void *buffer, int buflen);
+extern void Android_JNI_FlushCapturedAudio(void);
+extern void Android_JNI_CloseAudioDevice(const int iscapture);
+extern void Android_JNI_AudioSetThreadPriority(int iscapture, int device_id);
+
+/* Detecting device type */
+extern SDL_bool Android_IsDeXMode(void);
+extern SDL_bool Android_IsChromebook(void);
+
+#include "SDL_rwops.h"
+
+int Android_JNI_FileOpen(SDL_RWops* ctx, const char* fileName, const char* mode);
+Sint64 Android_JNI_FileSize(SDL_RWops* ctx);
+Sint64 Android_JNI_FileSeek(SDL_RWops* ctx, Sint64 offset, int whence);
+size_t Android_JNI_FileRead(SDL_RWops* ctx, void* buffer, size_t size, size_t maxnum);
+size_t Android_JNI_FileWrite(SDL_RWops* ctx, const void* buffer, size_t size, size_t num);
+int Android_JNI_FileClose(SDL_RWops* ctx);
+
+/* Environment support */
+void Android_JNI_GetManifestEnvironmentVariables(void);
+
+/* Clipboard support */
+int Android_JNI_SetClipboardText(const char *text);
+char *Android_JNI_GetClipboardText(void);
+SDL_bool Android_JNI_HasClipboardText(void);
+
+/* Power support */
+int Android_JNI_GetPowerInfo(int *plugged, int *charged, int *battery, int *seconds, int *percent);
+
+/* Joystick support */
+void Android_JNI_PollInputDevices(void);
+
+/* Haptic support */
+void Android_JNI_PollHapticDevices(void);
+void Android_JNI_HapticRun(int device_id, float intensity, int length);
+void Android_JNI_HapticStop(int device_id);
+
+/* Video */
+void Android_JNI_SuspendScreenSaver(SDL_bool suspend);
+
+/* Touch support */
+void Android_JNI_InitTouch(void);
+
+/* Threads */
+#include <jni.h>
+JNIEnv *Android_JNI_GetEnv(void);
+int Android_JNI_SetupThread(void);
+
+/* Locale */
+int Android_JNI_GetLocale(char *buf, size_t buflen);
+
+/* Generic messages */
+int Android_JNI_SendMessage(int command, int param);
+
+/* Init */
+JNIEXPORT void JNICALL SDL_Android_Init(JNIEnv *mEnv, jclass cls);
+
+/* MessageBox */
+#include "SDL_messagebox.h"
+int Android_JNI_ShowMessageBox(const SDL_MessageBoxData *messageboxdata, int *buttonid);
+
+/* Cursor support */
+int Android_JNI_CreateCustomCursor(SDL_Surface *surface, int hot_x, int hot_y);
+void Android_JNI_DestroyCustomCursor(int cursorID);
+SDL_bool Android_JNI_SetCustomCursor(int cursorID);
+SDL_bool Android_JNI_SetSystemCursor(int cursorID);
+
+/* Relative mouse support */
+SDL_bool Android_JNI_SupportsRelativeMouse(void);
+SDL_bool Android_JNI_SetRelativeMouseEnabled(SDL_bool enabled);
+
+/* Request permission */
+SDL_bool Android_JNI_RequestPermission(const char *permission);
+
+/* Show toast notification */
+int Android_JNI_ShowToast(const char *message, int duration, int gravity, int xOffset, int yOffset);
+
+int Android_JNI_OpenURL(const char *url);
+
+int SDL_GetAndroidSDKVersion(void);
+
+SDL_bool SDL_IsAndroidTablet(void);
+SDL_bool SDL_IsAndroidTV(void);
+SDL_bool SDL_IsChromebook(void);
+SDL_bool SDL_IsDeXMode(void);
+
+void Android_ActivityMutex_Lock(void);
+void Android_ActivityMutex_Unlock(void);
+void Android_ActivityMutex_Lock_Running(void);
+
+/* Ends C function definitions when using C++ */
+#ifdef __cplusplus
+/* *INDENT-OFF* */
+}
+/* *INDENT-ON* */
+#endif
+
+/* vi: set ts=4 sw=4 expandtab: */

third_party/SDL/src/core/freebsd/SDL_evdev_kbd_default_keyaccmap.h

@@ -0,0 +1,167 @@
+#include <sys/kbio.h>
+
+/* *INDENT-OFF* */ /* clang-format off */
+/*
+ * Automatically generated from /usr/share/vt/keymaps/us.acc.kbd.
+ * DO NOT EDIT!
+ */
+static keymap_t keymap_default_us_acc = { 0x6d, {
+/*                                                         alt
+ * scan                       cntrl          alt    alt   cntrl
+ * code  base   shift  cntrl  shift   alt   shift  cntrl  shift    spcl flgs
+ * ---------------------------------------------------------------------------
+ */
+/*00*/{{  NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP, }, 0xFF,0x00 },
+/*01*/{{ 0x1B,  0x1B,  0x1B,  0x1B,  0x1B,  0x1B,   DBG,   DBG, }, 0x03,0x00 },
+/*02*/{{  '1',   '!',   NOP,   NOP,   '1',   '!',   NOP,   NOP, }, 0x33,0x00 },
+/*03*/{{  '2',   '@',  0x00,  0x00,   '2',   '@',  0x00,  0x00, }, 0x00,0x00 },
+/*04*/{{  '3',   '#',   NOP,   NOP,   '3',   '#',   NOP,   NOP, }, 0x33,0x00 },
+/*05*/{{  '4',   '$',   NOP,   NOP,   '4',   '$',   NOP,   NOP, }, 0x33,0x00 },
+/*06*/{{  '5',   '%',   NOP,   NOP,   '5',   '%',   NOP,   NOP, }, 0x33,0x00 },
+/*07*/{{  '6',   '^',  0x1E,  0x1E,   '6',  DCIR,  0x1E,  0x1E, }, 0x04,0x00 },
+/*08*/{{  '7',   '&',   NOP,   NOP,   '7',   '&',   NOP,   NOP, }, 0x33,0x00 },
+/*09*/{{  '8',   '*',   NOP,   NOP,   '8',  DRIN,   NOP,   NOP, }, 0x37,0x00 },
+/*0a*/{{  '9',   '(',   NOP,   NOP,   '9',   '(',   NOP,   NOP, }, 0x33,0x00 },
+/*0b*/{{  '0',   ')',   NOP,   NOP,   '0',   ')',   NOP,   NOP, }, 0x33,0x00 },
+/*0c*/{{  '-',   '_',  0x1F,  0x1F,   '-',   '_',  0x1F,  0x1F, }, 0x00,0x00 },
+/*0d*/{{  '=',   '+',   NOP,   NOP,   '=',   '+',   NOP,   NOP, }, 0x33,0x00 },
+/*0e*/{{ 0x08,  0x08,  0x7F,  0x7F,  0x08,  0x08,  0x7F,  0x7F, }, 0x00,0x00 },
+/*0f*/{{ 0x09,  BTAB,  NEXT,  NEXT,  0x09,  BTAB,   NOP,   NOP, }, 0x77,0x00 },
+/*10*/{{  'q',   'Q',  0x11,  0x11,   'q',   'Q',  0x11,  0x11, }, 0x00,0x01 },
+/*11*/{{  'w',   'W',  0x17,  0x17,   'w',   'W',  0x17,  0x17, }, 0x00,0x01 },
+/*12*/{{  'e',   'E',  0x05,  0x05,   'e',   'E',  0x05,  0x05, }, 0x00,0x01 },
+/*13*/{{  'r',   'R',  0x12,  0x12,   'r',   'R',  0x12,  0x12, }, 0x00,0x01 },
+/*14*/{{  't',   'T',  0x14,  0x14,   't',   'T',  0x14,  0x14, }, 0x00,0x01 },
+/*15*/{{  'y',   'Y',  0x19,  0x19,   'y',   'Y',  0x19,  0x19, }, 0x00,0x01 },
+/*16*/{{  'u',   'U',  0x15,  0x15,   'u',   'U',  0x15,  0x15, }, 0x00,0x01 },
+/*17*/{{  'i',   'I',  0x09,  0x09,   'i',   'I',  0x09,  0x09, }, 0x00,0x01 },
+/*18*/{{  'o',   'O',  0x0F,  0x0F,   'o',   'O',  0x0F,  0x0F, }, 0x00,0x01 },
+/*19*/{{  'p',   'P',  0x10,  0x10,   'p',   'P',  0x10,  0x10, }, 0x00,0x01 },
+/*1a*/{{  '[',   '{',  0x1B,  0x1B,   '[',   '{',  0x1B,  0x1B, }, 0x00,0x00 },
+/*1b*/{{  ']',   '}',  0x1D,  0x1D,   ']',   '}',  0x1D,  0x1D, }, 0x00,0x00 },
+/*1c*/{{ 0x0D,  0x0D,  0x0A,  0x0A,  0x0D,  0x0D,  0x0A,  0x0A, }, 0x00,0x00 },
+/*1d*/{{ LCTR,  LCTR,  LCTR,  LCTR,  LCTR,  LCTR,  LCTR,  LCTR, }, 0xFF,0x00 },
+/*1e*/{{  'a',   'A',  0x01,  0x01,   'a',   'A',  0x01,  0x01, }, 0x00,0x01 },
+/*1f*/{{  's',   'S',  0x13,  0x13,   's',   'S',  0x13,  0x13, }, 0x00,0x01 },
+/*20*/{{  'd',   'D',  0x04,  0x04,   'd',   'D',  0x04,  0x04, }, 0x00,0x01 },
+/*21*/{{  'f',   'F',  0x06,  0x06,   'f',   'F',  0x06,  0x06, }, 0x00,0x01 },
+/*22*/{{  'g',   'G',  0x07,  0x07,   'g',   'G',  0x07,  0x07, }, 0x00,0x01 },
+/*23*/{{  'h',   'H',  0x08,  0x08,   'h',   'H',  0x08,  0x08, }, 0x00,0x01 },
+/*24*/{{  'j',   'J',  0x0A,  0x0A,   'j',   'J',  0x0A,  0x0A, }, 0x00,0x01 },
+/*25*/{{  'k',   'K',  0x0B,  0x0B,   'k',   'K',  0x0B,  0x0B, }, 0x00,0x01 },
+/*26*/{{  'l',   'L',  0x0C,  0x0C,   'l',   'L',  0x0C,  0x0C, }, 0x00,0x01 },
+/*27*/{{  ';',   ':',   NOP,   NOP,   ';',   ':',   NOP,   NOP, }, 0x33,0x00 },
+/*28*/{{ '\'',   '"',   NOP,   NOP,  DACU,  DUML,   NOP,   NOP, }, 0x3F,0x00 },
+/*29*/{{  '`',   '~',   NOP,   NOP,  DGRA,  DTIL,   NOP,   NOP, }, 0x3F,0x00 },
+/*2a*/{{  LSH,   LSH,   LSH,   LSH,   LSH,   LSH,   LSH,   LSH, }, 0xFF,0x00 },
+/*2b*/{{ '\\',   '|',  0x1C,  0x1C,  '\\',   '|',  0x1C,  0x1C, }, 0x00,0x00 },
+/*2c*/{{  'z',   'Z',  0x1A,  0x1A,   'z',   'Z',  0x1A,  0x1A, }, 0x00,0x01 },
+/*2d*/{{  'x',   'X',  0x18,  0x18,   'x',   'X',  0x18,  0x18, }, 0x00,0x01 },
+/*2e*/{{  'c',   'C',  0x03,  0x03,   'c',   'C',  0x03,  0x03, }, 0x00,0x01 },
+/*2f*/{{  'v',   'V',  0x16,  0x16,   'v',   'V',  0x16,  0x16, }, 0x00,0x01 },
+/*30*/{{  'b',   'B',  0x02,  0x02,   'b',   'B',  0x02,  0x02, }, 0x00,0x01 },
+/*31*/{{  'n',   'N',  0x0E,  0x0E,   'n',   'N',  0x0E,  0x0E, }, 0x00,0x01 },
+/*32*/{{  'm',   'M',  0x0D,  0x0D,   'm',   'M',  0x0D,  0x0D, }, 0x00,0x01 },
+/*33*/{{  ',',   '<',   NOP,   NOP,  DCED,   '<',   NOP,   NOP, }, 0x3B,0x00 },
+/*34*/{{  '.',   '>',   NOP,   NOP,   '.',   '>',   NOP,   NOP, }, 0x33,0x00 },
+/*35*/{{  '/',   '?',   NOP,   NOP,   '/',   '?',   NOP,   NOP, }, 0x33,0x00 },
+/*36*/{{  RSH,   RSH,   RSH,   RSH,   RSH,   RSH,   RSH,   RSH, }, 0xFF,0x00 },
+/*37*/{{  '*',   '*',   '*',   '*',   '*',   '*',   '*',   '*', }, 0x00,0x00 },
+/*38*/{{ LALT,  LALT,  LALT,  LALT,  LALT,  LALT,  LALT,  LALT, }, 0xFF,0x00 },
+/*39*/{{  ' ',   ' ',  0x00,  0x00,   ' ',   ' ',  SUSP,  SUSP, }, 0x03,0x00 },
+/*3a*/{{  CLK,   CLK,   CLK,   CLK,   CLK,   CLK,   CLK,   CLK, }, 0xFF,0x00 },
+/*3b*/{{ F( 1), F(13), F(25), F(37), S( 1), S(11), S( 1), S(11),}, 0xFF,0x00 },
+/*3c*/{{ F( 2), F(14), F(26), F(38), S( 2), S(12), S( 2), S(12),}, 0xFF,0x00 },
+/*3d*/{{ F( 3), F(15), F(27), F(39), S( 3), S(13), S( 3), S(13),}, 0xFF,0x00 },
+/*3e*/{{ F( 4), F(16), F(28), F(40), S( 4), S(14), S( 4), S(14),}, 0xFF,0x00 },
+/*3f*/{{ F( 5), F(17), F(29), F(41), S( 5), S(15), S( 5), S(15),}, 0xFF,0x00 },
+/*40*/{{ F( 6), F(18), F(30), F(42), S( 6), S(16), S( 6), S(16),}, 0xFF,0x00 },
+/*41*/{{ F( 7), F(19), F(31), F(43), S( 7), S( 7), S( 7), S( 7),}, 0xFF,0x00 },
+/*42*/{{ F( 8), F(20), F(32), F(44), S( 8), S( 8), S( 8), S( 8),}, 0xFF,0x00 },
+/*43*/{{ F( 9), F(21), F(33), F(45), S( 9), S( 9), S( 9), S( 9),}, 0xFF,0x00 },
+/*44*/{{ F(10), F(22), F(34), F(46), S(10), S(10), S(10), S(10),}, 0xFF,0x00 },
+/*45*/{{  NLK,   NLK,   NLK,   NLK,   NLK,   NLK,   NLK,   NLK, }, 0xFF,0x00 },
+/*46*/{{  SLK,   SLK,   SLK,   SLK,   SLK,   SLK,   SLK,   SLK, }, 0xFF,0x00 },
+/*47*/{{ F(49),  '7',   '7',   '7',   '7',   '7',   '7',   '7', }, 0x80,0x02 },
+/*48*/{{ F(50),  '8',   '8',   '8',   '8',   '8',   '8',   '8', }, 0x80,0x02 },
+/*49*/{{ F(51),  '9',   '9',   '9',   '9',   '9',   '9',   '9', }, 0x80,0x02 },
+/*4a*/{{ F(52),  '-',   '-',   '-',   '-',   '-',   '-',   '-', }, 0x80,0x02 },
+/*4b*/{{ F(53),  '4',   '4',   '4',   '4',   '4',   '4',   '4', }, 0x80,0x02 },
+/*4c*/{{ F(54),  '5',   '5',   '5',   '5',   '5',   '5',   '5', }, 0x80,0x02 },
+/*4d*/{{ F(55),  '6',   '6',   '6',   '6',   '6',   '6',   '6', }, 0x80,0x02 },
+/*4e*/{{ F(56),  '+',   '+',   '+',   '+',   '+',   '+',   '+', }, 0x80,0x02 },
+/*4f*/{{ F(57),  '1',   '1',   '1',   '1',   '1',   '1',   '1', }, 0x80,0x02 },
+/*50*/{{ F(58),  '2',   '2',   '2',   '2',   '2',   '2',   '2', }, 0x80,0x02 },
+/*51*/{{ F(59),  '3',   '3',   '3',   '3',   '3',   '3',   '3', }, 0x80,0x02 },
+/*52*/{{ F(60),  '0',   '0',   '0',   '0',   '0',   '0',   '0', }, 0x80,0x02 },
+/*53*/{{ 0x7F,   '.',   '.',   '.',   '.',   '.',   RBT,   RBT, }, 0x03,0x02 },
+/*54*/{{  NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP, }, 0xFF,0x00 },
+/*55*/{{  NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP, }, 0xFF,0x00 },
+/*56*/{{  NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP, }, 0xFF,0x00 },
+/*57*/{{ F(11), F(23), F(35), F(47), S(11), S(11), S(11), S(11),}, 0xFF,0x00 },
+/*58*/{{ F(12), F(24), F(36), F(48), S(12), S(12), S(12), S(12),}, 0xFF,0x00 },
+/*59*/{{ 0x0D,  0x0D,  0x0A,  0x0A,  0x0D,  0x0D,  0x0A,  0x0A, }, 0x00,0x00 },
+/*5a*/{{ RCTR,  RCTR,  RCTR,  RCTR,  RCTR,  RCTR,  RCTR,  RCTR, }, 0xFF,0x00 },
+/*5b*/{{  '/',   '/',   '/',   '/',   '/',   '/',   '/',   '/', }, 0x00,0x02 },
+/*5c*/{{ NEXT,  NEXT,   NOP,   NOP,   DBG,   DBG,   DBG,   DBG, }, 0xFF,0x00 },
+/*5d*/{{ RALT,  RALT,  RALT,  RALT,  RALT,  RALT,  RALT,  RALT, }, 0xFF,0x00 },
+/*5e*/{{ F(49), F(49), F(49), F(49), F(49), F(49), F(49), F(49),}, 0xFF,0x00 },
+/*5f*/{{ F(50), F(50), F(50), F(50), F(50), F(50), F(50), F(50),}, 0xFF,0x00 },
+/*60*/{{ F(51), F(51), F(51), F(51), F(51), F(51), F(51), F(51),}, 0xFF,0x00 },
+/*61*/{{ F(53), F(53), F(53), F(53), F(53), F(53), F(53), F(53),}, 0xFF,0x00 },
+/*62*/{{ F(55), F(55), F(55), F(55), F(55), F(55), F(55), F(55),}, 0xFF,0x00 },
+/*63*/{{ F(57), F(57), F(57), F(57), F(57), F(57), F(57), F(57),}, 0xFF,0x00 },
+/*64*/{{ F(58), F(58), F(58), F(58), F(58), F(58), F(58), F(58),}, 0xFF,0x00 },
+/*65*/{{ F(59), F(59), F(59), F(59), F(59), F(59), F(59), F(59),}, 0xFF,0x00 },
+/*66*/{{ F(60), F(60), F(60), F(60), F(60), F(60), F(60), F(60),}, 0xFF,0x00 },
+/*67*/{{ F(61), F(61), F(61), F(61), F(61), F(61),  RBT,  F(61),}, 0xFF,0x00 },
+/*68*/{{ SPSC,  SPSC,  SUSP,  SUSP,   NOP,   NOP,  SUSP,  SUSP, }, 0xFF,0x00 },
+/*69*/{{ F(62), F(62), F(62), F(62), F(62), F(62), F(62), F(62),}, 0xFF,0x00 },
+/*6a*/{{ F(63), F(63), F(63), F(63), F(63), F(63), F(63), F(63),}, 0xFF,0x00 },
+/*6b*/{{ F(64), F(64), F(64), F(64), F(64), F(64), F(64), F(64),}, 0xFF,0x00 },
+/*6c*/{{  NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP,   NOP, }, 0xFF,0x00 },
+} };
+
+static accentmap_t accentmap_default_us_acc = { 11, {
+    /* dgra=0 */
+    {  '`', { {  'a',0xe0 }, {  'A',0xc0 }, {  'e',0xe8 }, {  'E',0xc8 },
+	      {  'i',0xec }, {  'I',0xcc }, {  'o',0xf2 }, {  'O',0xd2 },
+	      {  'u',0xf9 }, {  'U',0xd9 }, }, },
+    /* dacu=1 */
+    { 0xb4, { {  'a',0xe1 }, {  'A',0xc1 }, {  'e',0xe9 }, {  'E',0xc9 },
+	      {  'i',0xed }, {  'I',0xcd }, {  'o',0xf3 }, {  'O',0xd3 },
+	      {  'u',0xfa }, {  'U',0xda }, {  'y',0xfd }, {  'Y',0xdd }, }, },
+    /* dcir=2 */
+    {  '^', { {  'a',0xe2 }, {  'A',0xc2 }, {  'e',0xea }, {  'E',0xca },
+	      {  'i',0xee }, {  'I',0xce }, {  'o',0xf4 }, {  'O',0xd4 },
+	      {  'u',0xfb }, {  'U',0xdb }, }, },
+    /* dtil=3 */
+    {  '~', { {  'a',0xe3 }, {  'A',0xc3 }, {  'n',0xf1 }, {  'N',0xd1 },
+	      {  'o',0xf5 }, {  'O',0xd5 }, }, },
+    /* dmac=4 */
+    { 0x00 }, 
+    /* dbre=5 */
+    { 0x00 }, 
+    /* ddot=6 */
+    { 0x00 }, 
+    /* duml=7 */
+    { 0xa8, { {  'a',0xe4 }, {  'A',0xc4 }, {  'e',0xeb }, {  'E',0xcb },
+	      {  'i',0xef }, {  'I',0xcf }, {  'o',0xf6 }, {  'O',0xd6 },
+	      {  'u',0xfc }, {  'U',0xdc }, {  'y',0xff }, }, },
+    /* dsla=8 */
+    { 0x00 }, 
+    /* drin=9 */
+    { 0xb0, { {  'a',0xe5 }, {  'A',0xc5 }, }, },
+    /* dced=10 */
+    { 0xb8, { {  'c',0xe7 }, {  'C',0xc7 }, }, },
+    /* dapo=11 */
+    { 0x00 }, 
+    /* ddac=12 */
+    { 0x00 }, 
+    /* dogo=13 */
+    { 0x00 }, 
+    /* dcar=14 */
+    { 0x00 }, 
+} };
+
+/* *INDENT-ON* */ /* clang-format on */

third_party/SDL/src/core/freebsd/SDL_evdev_kbd_freebsd.c

@@ -0,0 +1,601 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "../../SDL_internal.h"
+
+#include "../linux/SDL_evdev_kbd.h"
+#include "SDL_hints.h"
+
+#ifdef SDL_INPUT_FBSDKBIO
+
+/* This logic is adapted from drivers/tty/vt/keyboard.c in the Linux kernel source, slightly modified to work with FreeBSD */
+
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <sys/kbio.h>
+#include <sys/consio.h>
+
+#include <signal.h>
+
+#include "../../events/SDL_events_c.h"
+#include "SDL_evdev_kbd_default_keyaccmap.h"
+
+typedef void(fn_handler_fn)(SDL_EVDEV_keyboard_state *kbd);
+
+/*
+ * Keyboard State
+ */
+
+struct SDL_EVDEV_keyboard_state
+{
+    int console_fd;
+    int keyboard_fd;
+    unsigned long old_kbd_mode;
+    unsigned short **key_maps;
+    keymap_t *key_map;
+    keyboard_info_t *kbInfo;
+    unsigned char shift_down[4]; /* shift state counters.. */
+    SDL_bool dead_key_next;
+    int npadch; /* -1 or number assembled on pad */
+    accentmap_t *accents;
+    unsigned int diacr;
+    SDL_bool rep; /* flag telling character repeat */
+    unsigned char lockstate;
+    unsigned char ledflagstate;
+    char shift_state;
+    char text[128];
+    unsigned int text_len;
+};
+
+static int SDL_EVDEV_kbd_load_keymaps(SDL_EVDEV_keyboard_state *kbd)
+{
+    return ioctl(kbd->keyboard_fd, GIO_KEYMAP, kbd->key_map) >= 0;
+}
+
+static SDL_EVDEV_keyboard_state *kbd_cleanup_state = NULL;
+static int kbd_cleanup_sigactions_installed = 0;
+static int kbd_cleanup_atexit_installed = 0;
+
+static struct sigaction old_sigaction[NSIG];
+
+static int fatal_signals[] = {
+    /* Handlers for SIGTERM and SIGINT are installed in SDL_QuitInit. */
+    SIGHUP, SIGQUIT, SIGILL, SIGABRT,
+    SIGFPE, SIGSEGV, SIGPIPE, SIGBUS,
+    SIGSYS
+};
+
+static void kbd_cleanup(void)
+{
+    struct mouse_info mData;
+    SDL_EVDEV_keyboard_state *kbd = kbd_cleanup_state;
+    if (kbd == NULL) {
+        return;
+    }
+    kbd_cleanup_state = NULL;
+    SDL_zero(mData);
+    mData.operation = MOUSE_SHOW;
+    ioctl(kbd->keyboard_fd, KDSKBMODE, kbd->old_kbd_mode);
+    if (kbd->keyboard_fd != kbd->console_fd) {
+        close(kbd->keyboard_fd);
+    }
+    ioctl(kbd->console_fd, CONS_SETKBD, (unsigned long)(kbd->kbInfo->kb_index));
+    ioctl(kbd->console_fd, CONS_MOUSECTL, &mData);
+}
+
+void SDL_EVDEV_kbd_reraise_signal(int sig)
+{
+    raise(sig);
+}
+
+siginfo_t *SDL_EVDEV_kdb_cleanup_siginfo = NULL;
+void *SDL_EVDEV_kdb_cleanup_ucontext = NULL;
+
+static void kbd_cleanup_signal_action(int signum, siginfo_t *info, void *ucontext)
+{
+    struct sigaction *old_action_p = &(old_sigaction[signum]);
+    sigset_t sigset;
+
+    /* Restore original signal handler before going any further. */
+    sigaction(signum, old_action_p, NULL);
+
+    /* Unmask current signal. */
+    sigemptyset(&sigset);
+    sigaddset(&sigset, signum);
+    sigprocmask(SIG_UNBLOCK, &sigset, NULL);
+
+    /* Save original signal info and context for archeologists. */
+    SDL_EVDEV_kdb_cleanup_siginfo = info;
+    SDL_EVDEV_kdb_cleanup_ucontext = ucontext;
+
+    /* Restore keyboard. */
+    kbd_cleanup();
+
+    /* Reraise signal. */
+    SDL_EVDEV_kbd_reraise_signal(signum);
+}
+
+static void kbd_unregister_emerg_cleanup()
+{
+    int tabidx, signum;
+
+    kbd_cleanup_state = NULL;
+
+    if (!kbd_cleanup_sigactions_installed) {
+        return;
+    }
+    kbd_cleanup_sigactions_installed = 0;
+
+    for (tabidx = 0; tabidx < sizeof(fatal_signals) / sizeof(fatal_signals[0]); ++tabidx) {
+        struct sigaction *old_action_p;
+        struct sigaction cur_action;
+        signum = fatal_signals[tabidx];
+        old_action_p = &(old_sigaction[signum]);
+
+        /* Examine current signal action */
+        if (sigaction(signum, NULL, &cur_action)) {
+            continue;
+        }
+
+        /* Check if action installed and not modifed */
+        if (!(cur_action.sa_flags & SA_SIGINFO) || cur_action.sa_sigaction != &kbd_cleanup_signal_action) {
+            continue;
+        }
+
+        /* Restore original action */
+        sigaction(signum, old_action_p, NULL);
+    }
+}
+
+static void kbd_cleanup_atexit(void)
+{
+    /* Restore keyboard. */
+    kbd_cleanup();
+
+    /* Try to restore signal handlers in case shared library is being unloaded */
+    kbd_unregister_emerg_cleanup();
+}
+
+static void kbd_register_emerg_cleanup(SDL_EVDEV_keyboard_state *kbd)
+{
+    int tabidx, signum;
+
+    if (kbd_cleanup_state != NULL) {
+        return;
+    }
+    kbd_cleanup_state = kbd;
+
+    if (!kbd_cleanup_atexit_installed) {
+        /* Since glibc 2.2.3, atexit() (and on_exit(3)) can be used within a shared library to establish
+         * functions that are called when the shared library is unloaded.
+         * -- man atexit(3)
+         */
+        atexit(kbd_cleanup_atexit);
+        kbd_cleanup_atexit_installed = 1;
+    }
+
+    if (kbd_cleanup_sigactions_installed) {
+        return;
+    }
+    kbd_cleanup_sigactions_installed = 1;
+
+    for (tabidx = 0; tabidx < sizeof(fatal_signals) / sizeof(fatal_signals[0]); ++tabidx) {
+        struct sigaction *old_action_p;
+        struct sigaction new_action;
+        signum = fatal_signals[tabidx];
+        old_action_p = &(old_sigaction[signum]);
+        if (sigaction(signum, NULL, old_action_p)) {
+            continue;
+        }
+
+        /* Skip SIGHUP and SIGPIPE if handler is already installed
+         * - assume the handler will do the cleanup
+         */
+        if ((signum == SIGHUP || signum == SIGPIPE) && (old_action_p->sa_handler != SIG_DFL || (void (*)(int))old_action_p->sa_sigaction != SIG_DFL)) {
+            continue;
+        }
+
+        new_action = *old_action_p;
+        new_action.sa_flags |= SA_SIGINFO;
+        new_action.sa_sigaction = &kbd_cleanup_signal_action;
+        sigaction(signum, &new_action, NULL);
+    }
+}
+
+SDL_EVDEV_keyboard_state *SDL_EVDEV_kbd_init(void)
+{
+    SDL_EVDEV_keyboard_state *kbd;
+    struct mouse_info mData;
+    char flag_state;
+    char *devicePath;
+
+    SDL_zero(mData);
+    mData.operation = MOUSE_HIDE;
+    kbd = (SDL_EVDEV_keyboard_state *)SDL_calloc(1, sizeof(SDL_EVDEV_keyboard_state));
+    if (kbd == NULL) {
+        return NULL;
+    }
+
+    kbd->npadch = -1;
+
+    /* This might fail if we're not connected to a tty (e.g. on the Steam Link) */
+    kbd->keyboard_fd = kbd->console_fd = open("/dev/tty", O_RDONLY | O_CLOEXEC);
+
+    kbd->shift_state = 0;
+
+    kbd->accents = SDL_calloc(sizeof(accentmap_t), 1);
+    kbd->key_map = SDL_calloc(sizeof(keymap_t), 1);
+    kbd->kbInfo = SDL_calloc(sizeof(keyboard_info_t), 1);
+
+    ioctl(kbd->console_fd, KDGKBINFO, kbd->kbInfo);
+    ioctl(kbd->console_fd, CONS_MOUSECTL, &mData);
+
+    if (ioctl(kbd->console_fd, KDGKBSTATE, &flag_state) == 0) {
+        kbd->ledflagstate = flag_state;
+    }
+
+    if (ioctl(kbd->console_fd, GIO_DEADKEYMAP, kbd->accents) < 0) {
+        SDL_free(kbd->accents);
+        kbd->accents = &accentmap_default_us_acc;
+    }
+
+    if (ioctl(kbd->console_fd, KDGKBMODE, &kbd->old_kbd_mode) == 0) {
+        /* Set the keyboard in XLATE mode and load the keymaps */
+        ioctl(kbd->console_fd, KDSKBMODE, (unsigned long)(K_XLATE));
+        if (!SDL_EVDEV_kbd_load_keymaps(kbd)) {
+            SDL_free(kbd->key_map);
+            kbd->key_map = &keymap_default_us_acc;
+        }
+        /* Allow inhibiting keyboard mute with env. variable for debugging etc. */
+        if (SDL_getenv("SDL_INPUT_FREEBSD_KEEP_KBD") == NULL) {
+            /* Take keyboard from console and open the actual keyboard device.
+             * Ensures that the keystrokes do not leak through to the console.
+             */
+            ioctl(kbd->console_fd, CONS_RELKBD, 1ul);
+            SDL_asprintf(&devicePath, "/dev/kbd%d", kbd->kbInfo->kb_index);
+            kbd->keyboard_fd = open(devicePath, O_WRONLY | O_CLOEXEC);
+            if (kbd->keyboard_fd == -1) {
+                // Give keyboard back.
+                ioctl(kbd->console_fd, CONS_SETKBD, (unsigned long)(kbd->kbInfo->kb_index));
+                kbd->keyboard_fd = kbd->console_fd;
+            }
+
+            /* Make sure to restore keyboard if application fails to call
+             * SDL_Quit before exit or fatal signal is raised.
+             */
+            if (!SDL_GetHintBoolean(SDL_HINT_NO_SIGNAL_HANDLERS, SDL_FALSE)) {
+                kbd_register_emerg_cleanup(kbd);
+            }
+            SDL_free(devicePath);
+        } else
+            kbd->keyboard_fd = kbd->console_fd;
+    }
+
+    return kbd;
+}
+
+void SDL_EVDEV_kbd_quit(SDL_EVDEV_keyboard_state *kbd)
+{
+    struct mouse_info mData;
+
+    if (kbd == NULL) {
+        return;
+    }
+    SDL_zero(mData);
+    mData.operation = MOUSE_SHOW;
+    ioctl(kbd->console_fd, CONS_MOUSECTL, &mData);
+
+    kbd_unregister_emerg_cleanup();
+
+    if (kbd->keyboard_fd >= 0) {
+        /* Restore the original keyboard mode */
+        ioctl(kbd->keyboard_fd, KDSKBMODE, kbd->old_kbd_mode);
+
+        close(kbd->keyboard_fd);
+        if (kbd->console_fd != kbd->keyboard_fd && kbd->console_fd >= 0) {
+            // Give back keyboard.
+            ioctl(kbd->console_fd, CONS_SETKBD, (unsigned long)(kbd->kbInfo->kb_index));
+        }
+        kbd->console_fd = kbd->keyboard_fd = -1;
+    }
+
+    SDL_free(kbd);
+}
+
+/*
+ * Helper Functions.
+ */
+static void put_queue(SDL_EVDEV_keyboard_state *kbd, uint c)
+{
+    /* c is already part of a UTF-8 sequence and safe to add as a character */
+    if (kbd->text_len < (sizeof(kbd->text) - 1)) {
+        kbd->text[kbd->text_len++] = (char)c;
+    }
+}
+
+static void put_utf8(SDL_EVDEV_keyboard_state *kbd, uint c)
+{
+    if (c < 0x80)
+        /*  0******* */
+        put_queue(kbd, c);
+    else if (c < 0x800) {
+        /* 110***** 10****** */
+        put_queue(kbd, 0xc0 | (c >> 6));
+        put_queue(kbd, 0x80 | (c & 0x3f));
+    } else if (c < 0x10000) {
+        if (c >= 0xD800 && c < 0xE000) {
+            return;
+        }
+        if (c == 0xFFFF) {
+            return;
+        }
+        /* 1110**** 10****** 10****** */
+        put_queue(kbd, 0xe0 | (c >> 12));
+        put_queue(kbd, 0x80 | ((c >> 6) & 0x3f));
+        put_queue(kbd, 0x80 | (c & 0x3f));
+    } else if (c < 0x110000) {
+        /* 11110*** 10****** 10****** 10****** */
+        put_queue(kbd, 0xf0 | (c >> 18));
+        put_queue(kbd, 0x80 | ((c >> 12) & 0x3f));
+        put_queue(kbd, 0x80 | ((c >> 6) & 0x3f));
+        put_queue(kbd, 0x80 | (c & 0x3f));
+    }
+}
+
+/*
+ * We have a combining character DIACR here, followed by the character CH.
+ * If the combination occurs in the table, return the corresponding value.
+ * Otherwise, if CH is a space or equals DIACR, return DIACR.
+ * Otherwise, conclude that DIACR was not combining after all,
+ * queue it and return CH.
+ */
+static unsigned int handle_diacr(SDL_EVDEV_keyboard_state *kbd, unsigned int ch)
+{
+    unsigned int d = kbd->diacr;
+    unsigned int i, j;
+
+    kbd->diacr = 0;
+
+    for (i = 0; i < kbd->accents->n_accs; i++) {
+        if (kbd->accents->acc[i].accchar == d) {
+            for (j = 0; j < NUM_ACCENTCHARS; ++j) {
+                if (kbd->accents->acc[i].map[j][0] == 0) { /* end of table */
+                    break;
+                }
+                if (kbd->accents->acc[i].map[j][0] == ch) {
+                    return kbd->accents->acc[i].map[j][1];
+                }
+            }
+        }
+    }
+
+    if (ch == ' ' || ch == d) {
+        put_utf8(kbd, d);
+        return 0;
+    }
+    put_utf8(kbd, d);
+
+    return ch;
+}
+
+static int vc_kbd_led(SDL_EVDEV_keyboard_state *kbd, int flag)
+{
+    return (kbd->ledflagstate & flag) != 0;
+}
+
+static void chg_vc_kbd_led(SDL_EVDEV_keyboard_state *kbd, int flag)
+{
+    kbd->ledflagstate ^= flag;
+    ioctl(kbd->keyboard_fd, KDSKBSTATE, (unsigned long)(kbd->ledflagstate));
+}
+
+/*
+ * Special function handlers
+ */
+
+static void k_self(SDL_EVDEV_keyboard_state *kbd, unsigned int value, char up_flag)
+{
+    if (up_flag) {
+        return; /* no action, if this is a key release */
+    }
+
+    if (kbd->diacr) {
+        value = handle_diacr(kbd, value);
+    }
+
+    if (kbd->dead_key_next) {
+        kbd->dead_key_next = SDL_FALSE;
+        kbd->diacr = value;
+        return;
+    }
+    put_utf8(kbd, value);
+}
+
+static void k_deadunicode(SDL_EVDEV_keyboard_state *kbd, unsigned int value, char up_flag)
+{
+    if (up_flag)
+        return;
+
+    kbd->diacr = (kbd->diacr ? handle_diacr(kbd, value) : value);
+}
+
+static void k_shift(SDL_EVDEV_keyboard_state *kbd, unsigned char value, char up_flag)
+{
+    int old_state = kbd->shift_state;
+
+    if (kbd->rep)
+        return;
+
+    if (up_flag) {
+        /*
+         * handle the case that two shift or control
+         * keys are depressed simultaneously
+         */
+        if (kbd->shift_down[value]) {
+            kbd->shift_down[value]--;
+        }
+    } else
+        kbd->shift_down[value]++;
+
+    if (kbd->shift_down[value])
+        kbd->shift_state |= (1 << value);
+    else
+        kbd->shift_state &= ~(1 << value);
+
+    /* kludge */
+    if (up_flag && kbd->shift_state != old_state && kbd->npadch != -1) {
+        put_utf8(kbd, kbd->npadch);
+        kbd->npadch = -1;
+    }
+}
+
+void SDL_EVDEV_kbd_keycode(SDL_EVDEV_keyboard_state *kbd, unsigned int keycode, int down)
+{
+    keymap_t key_map;
+    struct keyent_t keysym;
+    unsigned int final_key_state;
+    unsigned int map_from_key_sym;
+
+    if (kbd == NULL) {
+        return;
+    }
+
+    key_map = *kbd->key_map;
+
+    kbd->rep = (down == 2);
+
+    if (keycode < NUM_KEYS) {
+        if (keycode >= 89 && keycode <= 95) {
+            /* These constitute unprintable language-related keys, so ignore them. */
+            return;
+        }
+        if (keycode > 95) {
+            keycode -= 7;
+        }
+        if (vc_kbd_led(kbd, ALKED) || (kbd->shift_state & 0x8)) {
+            keycode += ALTGR_OFFSET;
+        }
+        keysym = key_map.key[keycode];
+    } else {
+        return;
+    }
+
+    final_key_state = kbd->shift_state & 0x7;
+    if ((keysym.flgs & FLAG_LOCK_C) && vc_kbd_led(kbd, LED_CAP)) {
+        final_key_state ^= 0x1;
+    }
+    if ((keysym.flgs & FLAG_LOCK_N) && vc_kbd_led(kbd, LED_NUM)) {
+        final_key_state ^= 0x1;
+    }
+
+    map_from_key_sym = keysym.map[final_key_state];
+    if ((keysym.spcl & (0x80 >> final_key_state)) || (map_from_key_sym & SPCLKEY)) {
+        /* Special function.*/
+        if (map_from_key_sym == 0)
+            return; /* Nothing to do. */
+        if (map_from_key_sym & SPCLKEY) {
+            map_from_key_sym &= ~SPCLKEY;
+        }
+        if (map_from_key_sym >= F_ACC && map_from_key_sym <= L_ACC) {
+            /* Accent function.*/
+            unsigned int accent_index = map_from_key_sym - F_ACC;
+            if (kbd->accents->acc[accent_index].accchar != 0) {
+                k_deadunicode(kbd, kbd->accents->acc[accent_index].accchar, !down);
+            }
+        } else {
+            switch (map_from_key_sym) {
+            case ASH: /* alt/meta shift */
+                k_shift(kbd, 3, down == 0);
+                break;
+            case LSHA: /* left shift + alt lock */
+            case RSHA: /* right shift + alt lock */
+                if (down == 0) {
+                    chg_vc_kbd_led(kbd, ALKED);
+                }
+            case LSH: /* left shift */
+            case RSH: /* right shift */
+                k_shift(kbd, 0, down == 0);
+                break;
+            case LCTRA: /* left ctrl + alt lock */
+            case RCTRA: /* right ctrl + alt lock */
+                if (down == 0) {
+                    chg_vc_kbd_led(kbd, ALKED);
+                }
+            case LCTR: /* left ctrl */
+            case RCTR: /* right ctrl */
+                k_shift(kbd, 1, down == 0);
+                break;
+            case LALTA: /* left alt + alt lock */
+            case RALTA: /* right alt + alt lock */
+                if (down == 0) {
+                    chg_vc_kbd_led(kbd, ALKED);
+                }
+            case LALT: /* left alt */
+            case RALT: /* right alt */
+                k_shift(kbd, 2, down == 0);
+                break;
+            case ALK: /* alt lock */
+                if (down == 1) {
+                    chg_vc_kbd_led(kbd, ALKED);
+                }
+                break;
+            case CLK: /* caps lock*/
+                if (down == 1) {
+                    chg_vc_kbd_led(kbd, CLKED);
+                }
+                break;
+            case NLK: /* num lock */
+                if (down == 1) {
+                    chg_vc_kbd_led(kbd, NLKED);
+                }
+                break;
+            case SLK: /* scroll lock */
+                if (down == 1) {
+                    chg_vc_kbd_led(kbd, SLKED);
+                }
+                break;
+            default:
+                return;
+            }
+        }
+    } else {
+        if (map_from_key_sym == '\n' || map_from_key_sym == '\r') {
+            if (kbd->diacr) {
+                kbd->diacr = 0;
+                return;
+            }
+        }
+        if (map_from_key_sym >= ' ' && map_from_key_sym != 127) {
+            k_self(kbd, map_from_key_sym, !down);
+        }
+    }
+
+    if (kbd->text_len > 0) {
+        kbd->text[kbd->text_len] = '\0';
+        SDL_SendKeyboardText(kbd->text);
+        kbd->text_len = 0;
+    }
+}
+
+#endif /* SDL_INPUT_FBSDKBIO */
+
+/* vi: set ts=4 sw=4 expandtab: */