alphatau 0.1.0

data/ext/alphatau/native/sdl/src/audio/SDL_audio.c

@@ -0,0 +1,1647 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2018 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"
+
+/* Allow access to a raw mixing buffer */
+
+#include "SDL.h"
+#include "SDL_audio.h"
+#include "SDL_audio_c.h"
+#include "SDL_sysaudio.h"
+#include "../thread/SDL_systhread.h"
+
+#define _THIS SDL_AudioDevice *_this
+
+static SDL_AudioDriver current_audio;
+static SDL_AudioDevice *open_devices[16];
+
+/* Available audio drivers */
+static const AudioBootStrap *const bootstrap[] = {
+#if SDL_AUDIO_DRIVER_PULSEAUDIO
+    &PULSEAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ALSA
+    &ALSA_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_SNDIO
+    &SNDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_NETBSD
+    &NETBSDAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_OSS
+    &DSP_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_QSA
+    &QSAAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_SUNAUDIO
+    &SUNAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ARTS
+    &ARTS_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ESD
+    &ESD_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_NACL
+    &NACLAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_NAS
+    &NAS_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_WASAPI
+    &WASAPI_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DSOUND
+    &DSOUND_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_WINMM
+    &WINMM_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_PAUDIO
+    &PAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_HAIKU
+    &HAIKUAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_COREAUDIO
+    &COREAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_FUSIONSOUND
+    &FUSIONSOUND_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ANDROID
+    &ANDROIDAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_PSP
+    &PSPAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_EMSCRIPTEN
+    &EMSCRIPTENAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_JACK
+    &JACK_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DISK
+    &DISKAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DUMMY
+    &DUMMYAUDIO_bootstrap,
+#endif
+    NULL
+};
+
+
+#ifdef HAVE_LIBSAMPLERATE_H
+#ifdef SDL_LIBSAMPLERATE_DYNAMIC
+static void *SRC_lib = NULL;
+#endif
+SDL_bool SRC_available = SDL_FALSE;
+int SRC_converter = 0;
+SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error) = NULL;
+int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data) = NULL;
+int (*SRC_src_reset)(SRC_STATE *state) = NULL;
+SRC_STATE* (*SRC_src_delete)(SRC_STATE *state) = NULL;
+const char* (*SRC_src_strerror)(int error) = NULL;
+
+static SDL_bool
+LoadLibSampleRate(void)
+{
+    const char *hint = SDL_GetHint(SDL_HINT_AUDIO_RESAMPLING_MODE);
+
+    SRC_available = SDL_FALSE;
+    SRC_converter = 0;
+
+    if (!hint || *hint == '0' || SDL_strcasecmp(hint, "default") == 0) {
+        return SDL_FALSE;  /* don't load anything. */
+    } else if (*hint == '1' || SDL_strcasecmp(hint, "fast") == 0) {
+        SRC_converter = SRC_SINC_FASTEST;
+    } else if (*hint == '2' || SDL_strcasecmp(hint, "medium") == 0) {
+        SRC_converter = SRC_SINC_MEDIUM_QUALITY;
+    } else if (*hint == '3' || SDL_strcasecmp(hint, "best") == 0) {
+        SRC_converter = SRC_SINC_BEST_QUALITY;
+    } else {
+        return SDL_FALSE;  /* treat it like "default", don't load anything. */
+    }
+
+#ifdef SDL_LIBSAMPLERATE_DYNAMIC
+    SDL_assert(SRC_lib == NULL);
+    SRC_lib = SDL_LoadObject(SDL_LIBSAMPLERATE_DYNAMIC);
+    if (!SRC_lib) {
+        SDL_ClearError();
+        return SDL_FALSE;
+    }
+
+    SRC_src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(SRC_lib, "src_new");
+    SRC_src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(SRC_lib, "src_process");
+    SRC_src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(SRC_lib, "src_reset");
+    SRC_src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(SRC_lib, "src_delete");
+    SRC_src_strerror = (const char* (*)(int error))SDL_LoadFunction(SRC_lib, "src_strerror");
+
+    if (!SRC_src_new || !SRC_src_process || !SRC_src_reset || !SRC_src_delete || !SRC_src_strerror) {
+        SDL_UnloadObject(SRC_lib);
+        SRC_lib = NULL;
+        return SDL_FALSE;
+    }
+#else
+    SRC_src_new = src_new;
+    SRC_src_process = src_process;
+    SRC_src_reset = src_reset;
+    SRC_src_delete = src_delete;
+    SRC_src_strerror = src_strerror;
+#endif
+
+    SRC_available = SDL_TRUE;
+    return SDL_TRUE;
+}
+
+static void
+UnloadLibSampleRate(void)
+{
+#ifdef SDL_LIBSAMPLERATE_DYNAMIC
+    if (SRC_lib != NULL) {
+        SDL_UnloadObject(SRC_lib);
+    }
+    SRC_lib = NULL;
+#endif
+
+    SRC_available = SDL_FALSE;
+    SRC_src_new = NULL;
+    SRC_src_process = NULL;
+    SRC_src_reset = NULL;
+    SRC_src_delete = NULL;
+    SRC_src_strerror = NULL;
+}
+#endif
+
+static SDL_AudioDevice *
+get_audio_device(SDL_AudioDeviceID id)
+{
+    id--;
+    if ((id >= SDL_arraysize(open_devices)) || (open_devices[id] == NULL)) {
+        SDL_SetError("Invalid audio device ID");
+        return NULL;
+    }
+
+    return open_devices[id];
+}
+
+
+/* stubs for audio drivers that don't need a specific entry point... */
+static void
+SDL_AudioDetectDevices_Default(void)
+{
+    /* you have to write your own implementation if these assertions fail. */
+    SDL_assert(current_audio.impl.OnlyHasDefaultOutputDevice);
+    SDL_assert(current_audio.impl.OnlyHasDefaultCaptureDevice || !current_audio.impl.HasCaptureSupport);
+
+    SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, (void *) ((size_t) 0x1));
+    if (current_audio.impl.HasCaptureSupport) {
+        SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, (void *) ((size_t) 0x2));
+    }
+}
+
+static void
+SDL_AudioThreadInit_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioThreadDeinit_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioBeginLoopIteration_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioWaitDevice_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioPlayDevice_Default(_THIS)
+{                               /* no-op. */
+}
+
+static int
+SDL_AudioGetPendingBytes_Default(_THIS)
+{
+    return 0;
+}
+
+static Uint8 *
+SDL_AudioGetDeviceBuf_Default(_THIS)
+{
+    return NULL;
+}
+
+static int
+SDL_AudioCaptureFromDevice_Default(_THIS, void *buffer, int buflen)
+{
+    return -1;  /* just fail immediately. */
+}
+
+static void
+SDL_AudioFlushCapture_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioPrepareToClose_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioCloseDevice_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioDeinitialize_Default(void)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioFreeDeviceHandle_Default(void *handle)
+{                               /* no-op. */
+}
+
+
+static int
+SDL_AudioOpenDevice_Default(_THIS, void *handle, const char *devname, int iscapture)
+{
+    return SDL_Unsupported();
+}
+
+static SDL_INLINE SDL_bool
+is_in_audio_device_thread(SDL_AudioDevice * device)
+{
+    /* The device thread locks the same mutex, but not through the public API.
+       This check is in case the application, in the audio callback,
+       tries to lock the thread that we've already locked from the
+       device thread...just in case we only have non-recursive mutexes. */
+    if (device->thread && (SDL_ThreadID() == device->threadid)) {
+        return SDL_TRUE;
+    }
+
+    return SDL_FALSE;
+}
+
+static void
+SDL_AudioLockDevice_Default(SDL_AudioDevice * device)
+{
+    if (!is_in_audio_device_thread(device)) {
+        SDL_LockMutex(device->mixer_lock);
+    }
+}
+
+static void
+SDL_AudioUnlockDevice_Default(SDL_AudioDevice * device)
+{
+    if (!is_in_audio_device_thread(device)) {
+        SDL_UnlockMutex(device->mixer_lock);
+    }
+}
+
+static void
+SDL_AudioLockOrUnlockDeviceWithNoMixerLock(SDL_AudioDevice * device)
+{
+}
+
+static void
+finish_audio_entry_points_init(void)
+{
+    /*
+     * Fill in stub functions for unused driver entry points. This lets us
+     *  blindly call them without having to check for validity first.
+     */
+
+    if (current_audio.impl.SkipMixerLock) {
+        if (current_audio.impl.LockDevice == NULL) {
+            current_audio.impl.LockDevice = SDL_AudioLockOrUnlockDeviceWithNoMixerLock;
+        }
+        if (current_audio.impl.UnlockDevice == NULL) {
+            current_audio.impl.UnlockDevice = SDL_AudioLockOrUnlockDeviceWithNoMixerLock;
+        }
+    }
+
+#define FILL_STUB(x) \
+        if (current_audio.impl.x == NULL) { \
+            current_audio.impl.x = SDL_Audio##x##_Default; \
+        }
+    FILL_STUB(DetectDevices);
+    FILL_STUB(OpenDevice);
+    FILL_STUB(ThreadInit);
+    FILL_STUB(ThreadDeinit);
+    FILL_STUB(BeginLoopIteration);
+    FILL_STUB(WaitDevice);
+    FILL_STUB(PlayDevice);
+    FILL_STUB(GetPendingBytes);
+    FILL_STUB(GetDeviceBuf);
+    FILL_STUB(CaptureFromDevice);
+    FILL_STUB(FlushCapture);
+    FILL_STUB(PrepareToClose);
+    FILL_STUB(CloseDevice);
+    FILL_STUB(LockDevice);
+    FILL_STUB(UnlockDevice);
+    FILL_STUB(FreeDeviceHandle);
+    FILL_STUB(Deinitialize);
+#undef FILL_STUB
+}
+
+
+/* device hotplug support... */
+
+static int
+add_audio_device(const char *name, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
+{
+    int retval = -1;
+    const size_t size = sizeof (SDL_AudioDeviceItem) + SDL_strlen(name) + 1;
+    SDL_AudioDeviceItem *item = (SDL_AudioDeviceItem *) SDL_malloc(size);
+    if (item == NULL) {
+        return -1;
+    }
+
+    SDL_assert(handle != NULL);  /* we reserve NULL, audio backends can't use it. */
+
+    item->handle = handle;
+    SDL_strlcpy(item->name, name, size - sizeof (SDL_AudioDeviceItem));
+
+    SDL_LockMutex(current_audio.detectionLock);
+    item->next = *devices;
+    *devices = item;
+    retval = (*devCount)++;
+    SDL_UnlockMutex(current_audio.detectionLock);
+
+    return retval;
+}
+
+static SDL_INLINE int
+add_capture_device(const char *name, void *handle)
+{
+    SDL_assert(current_audio.impl.HasCaptureSupport);
+    return add_audio_device(name, handle, &current_audio.inputDevices, &current_audio.inputDeviceCount);
+}
+
+static SDL_INLINE int
+add_output_device(const char *name, void *handle)
+{
+    return add_audio_device(name, handle, &current_audio.outputDevices, &current_audio.outputDeviceCount);
+}
+
+static void
+free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
+{
+    SDL_AudioDeviceItem *item, *next;
+    for (item = *devices; item != NULL; item = next) {
+        next = item->next;
+        if (item->handle != NULL) {
+            current_audio.impl.FreeDeviceHandle(item->handle);
+        }
+        SDL_free(item);
+    }
+    *devices = NULL;
+    *devCount = 0;
+}
+
+
+/* The audio backends call this when a new device is plugged in. */
+void
+SDL_AddAudioDevice(const int iscapture, const char *name, void *handle)
+{
+    const int device_index = iscapture ? add_capture_device(name, handle) : add_output_device(name, handle);
+    if (device_index != -1) {
+        /* Post the event, if desired */
+        if (SDL_GetEventState(SDL_AUDIODEVICEADDED) == SDL_ENABLE) {
+            SDL_Event event;
+            SDL_zero(event);
+            event.adevice.type = SDL_AUDIODEVICEADDED;
+            event.adevice.which = device_index;
+            event.adevice.iscapture = iscapture;
+            SDL_PushEvent(&event);
+        }
+    }
+}
+
+/* The audio backends call this when a currently-opened device is lost. */
+void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
+{
+    SDL_assert(get_audio_device(device->id) == device);
+
+    if (!SDL_AtomicGet(&device->enabled)) {
+        return;  /* don't report disconnects more than once. */
+    }
+
+    if (SDL_AtomicGet(&device->shutdown)) {
+        return;  /* don't report disconnect if we're trying to close device. */
+    }
+
+    /* Ends the audio callback and mark the device as STOPPED, but the
+       app still needs to close the device to free resources. */
+    current_audio.impl.LockDevice(device);
+    SDL_AtomicSet(&device->enabled, 0);
+    current_audio.impl.UnlockDevice(device);
+
+    /* Post the event, if desired */
+    if (SDL_GetEventState(SDL_AUDIODEVICEREMOVED) == SDL_ENABLE) {
+        SDL_Event event;
+        SDL_zero(event);
+        event.adevice.type = SDL_AUDIODEVICEREMOVED;
+        event.adevice.which = device->id;
+        event.adevice.iscapture = device->iscapture ? 1 : 0;
+        SDL_PushEvent(&event);
+    }
+}
+
+static void
+mark_device_removed(void *handle, SDL_AudioDeviceItem *devices, SDL_bool *removedFlag)
+{
+    SDL_AudioDeviceItem *item;
+    SDL_assert(handle != NULL);
+    for (item = devices; item != NULL; item = item->next) {
+        if (item->handle == handle) {
+            item->handle = NULL;
+            *removedFlag = SDL_TRUE;
+            return;
+        }
+    }
+}
+
+/* The audio backends call this when a device is removed from the system. */
+void
+SDL_RemoveAudioDevice(const int iscapture, void *handle)
+{
+    int device_index;
+    SDL_AudioDevice *device = NULL;
+
+    SDL_LockMutex(current_audio.detectionLock);
+    if (iscapture) {
+        mark_device_removed(handle, current_audio.inputDevices, &current_audio.captureDevicesRemoved);
+    } else {
+        mark_device_removed(handle, current_audio.outputDevices, &current_audio.outputDevicesRemoved);
+    }
+    for (device_index = 0; device_index < SDL_arraysize(open_devices); device_index++)
+    {
+        device = open_devices[device_index];
+        if (device != NULL && device->handle == handle)
+        {
+            SDL_OpenedAudioDeviceDisconnected(device);
+            break;
+        }
+    }
+    SDL_UnlockMutex(current_audio.detectionLock);
+
+    current_audio.impl.FreeDeviceHandle(handle);
+}
+
+
+
+/* buffer queueing support... */
+
+static void SDLCALL
+SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
+{
+    /* this function always holds the mixer lock before being called. */
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
+    size_t dequeued;
+
+    SDL_assert(device != NULL);  /* this shouldn't ever happen, right?! */
+    SDL_assert(!device->iscapture);  /* this shouldn't ever happen, right?! */
+    SDL_assert(len >= 0);  /* this shouldn't ever happen, right?! */
+
+    dequeued = SDL_ReadFromDataQueue(device->buffer_queue, stream, len);
+    stream += dequeued;
+    len -= (int) dequeued;
+
+    if (len > 0) {  /* fill any remaining space in the stream with silence. */
+        SDL_assert(SDL_CountDataQueue(device->buffer_queue) == 0);
+        SDL_memset(stream, device->spec.silence, len);
+    }
+}
+
+static void SDLCALL
+SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
+{
+    /* this function always holds the mixer lock before being called. */
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
+
+    SDL_assert(device != NULL);  /* this shouldn't ever happen, right?! */
+    SDL_assert(device->iscapture);  /* this shouldn't ever happen, right?! */
+    SDL_assert(len >= 0);  /* this shouldn't ever happen, right?! */
+
+    /* note that if this needs to allocate more space and run out of memory,
+       we have no choice but to quietly drop the data and hope it works out
+       later, but you probably have bigger problems in this case anyhow. */
+    SDL_WriteToDataQueue(device->buffer_queue, stream, len);
+}
+
+int
+SDL_QueueAudio(SDL_AudioDeviceID devid, const void *data, Uint32 len)
+{
+    SDL_AudioDevice *device = get_audio_device(devid);
+    int rc = 0;
+
+    if (!device) {
+        return -1;  /* get_audio_device() will have set the error state */
+    } else if (device->iscapture) {
+        return SDL_SetError("This is a capture device, queueing not allowed");
+    } else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
+        return SDL_SetError("Audio device has a callback, queueing not allowed");
+    }
+
+    if (len > 0) {
+        current_audio.impl.LockDevice(device);
+        rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
+        current_audio.impl.UnlockDevice(device);
+    }
+
+    return rc;
+}
+
+Uint32
+SDL_DequeueAudio(SDL_AudioDeviceID devid, void *data, Uint32 len)
+{
+    SDL_AudioDevice *device = get_audio_device(devid);
+    Uint32 rc;
+
+    if ( (len == 0) ||  /* nothing to do? */
+         (!device) ||  /* called with bogus device id */
+         (!device->iscapture) ||  /* playback devices can't dequeue */
+         (device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
+        return 0;  /* just report zero bytes dequeued. */
+    }
+
+    current_audio.impl.LockDevice(device);
+    rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
+    current_audio.impl.UnlockDevice(device);
+    return rc;
+}
+
+Uint32
+SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
+{
+    Uint32 retval = 0;
+    SDL_AudioDevice *device = get_audio_device(devid);
+
+    if (!device) {
+        return 0;
+    }
+
+    /* Nothing to do unless we're set up for queueing. */
+    if (device->callbackspec.callback == SDL_BufferQueueDrainCallback) {
+        current_audio.impl.LockDevice(device);
+        retval = ((Uint32) SDL_CountDataQueue(device->buffer_queue)) + current_audio.impl.GetPendingBytes(device);
+        current_audio.impl.UnlockDevice(device);
+    } else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
+        current_audio.impl.LockDevice(device);
+        retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
+        current_audio.impl.UnlockDevice(device);
+    }
+
+    return retval;
+}
+
+void
+SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
+{
+    SDL_AudioDevice *device = get_audio_device(devid);
+
+    if (!device) {
+        return;  /* nothing to do. */
+    }
+
+    /* Blank out the device and release the mutex. Free it afterwards. */
+    current_audio.impl.LockDevice(device);
+
+    /* Keep up to two packets in the pool to reduce future malloc pressure. */
+    SDL_ClearDataQueue(device->buffer_queue, SDL_AUDIOBUFFERQUEUE_PACKETLEN * 2);
+
+    current_audio.impl.UnlockDevice(device);
+}
+
+
+/* The general mixing thread function */
+static int SDLCALL
+SDL_RunAudio(void *devicep)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
+    void *udata = device->callbackspec.userdata;
+    SDL_AudioCallback callback = device->callbackspec.callback;
+    int data_len = 0;
+    Uint8 *data;
+
+    SDL_assert(!device->iscapture);
+
+    /* The audio mixing is always a high priority thread */
+    SDL_SetThreadPriority(SDL_THREAD_PRIORITY_TIME_CRITICAL);
+
+    /* Perform any thread setup */
+    device->threadid = SDL_ThreadID();
+    current_audio.impl.ThreadInit(device);
+
+    /* Loop, filling the audio buffers */
+    while (!SDL_AtomicGet(&device->shutdown)) {
+        current_audio.impl.BeginLoopIteration(device);
+        data_len = device->callbackspec.size;
+
+        /* Fill the current buffer with sound */
+        if (!device->stream && SDL_AtomicGet(&device->enabled)) {
+            SDL_assert(data_len == device->spec.size);
+            data = current_audio.impl.GetDeviceBuf(device);
+        } else {
+            /* if the device isn't enabled, we still write to the
+               work_buffer, so the app's callback will fire with
+               a regular frequency, in case they depend on that
+               for timing or progress. They can use hotplug
+               now to know if the device failed.
+               Streaming playback uses work_buffer, too. */
+            data = NULL;
+        }
+
+        if (data == NULL) {
+            data = device->work_buffer;
+        }
+
+        /* !!! FIXME: this should be LockDevice. */
+        SDL_LockMutex(device->mixer_lock);
+        if (SDL_AtomicGet(&device->paused)) {
+            SDL_memset(data, device->spec.silence, data_len);
+        } else {
+            callback(udata, data, data_len);
+        }
+        SDL_UnlockMutex(device->mixer_lock);
+
+        if (device->stream) {
+            /* Stream available audio to device, converting/resampling. */
+            /* if this fails...oh well. We'll play silence here. */
+            SDL_AudioStreamPut(device->stream, data, data_len);
+
+            while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->spec.size)) {
+                int got;
+                data = SDL_AtomicGet(&device->enabled) ? current_audio.impl.GetDeviceBuf(device) : NULL;
+                got = SDL_AudioStreamGet(device->stream, data ? data : device->work_buffer, device->spec.size);
+                SDL_assert((got < 0) || (got == device->spec.size));
+
+                if (data == NULL) {  /* device is having issues... */
+                    const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
+                    SDL_Delay(delay);  /* wait for as long as this buffer would have played. Maybe device recovers later? */
+                } else {
+                    if (got != device->spec.size) {
+                        SDL_memset(data, device->spec.silence, device->spec.size);
+                    }
+                    current_audio.impl.PlayDevice(device);
+                    current_audio.impl.WaitDevice(device);
+                }
+            }
+        } else if (data == device->work_buffer) {
+            /* nothing to do; pause like we queued a buffer to play. */
+            const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
+            SDL_Delay(delay);
+        } else {  /* writing directly to the device. */
+            /* queue this buffer and wait for it to finish playing. */
+            current_audio.impl.PlayDevice(device);
+            current_audio.impl.WaitDevice(device);
+        }
+    }
+
+    current_audio.impl.PrepareToClose(device);
+
+    /* Wait for the audio to drain. */
+    SDL_Delay(((device->spec.samples * 1000) / device->spec.freq) * 2);
+
+    current_audio.impl.ThreadDeinit(device);
+
+    return 0;
+}
+
+/* !!! FIXME: this needs to deal with device spec changes. */
+/* The general capture thread function */
+static int SDLCALL
+SDL_CaptureAudio(void *devicep)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
+    const int silence = (int) device->spec.silence;
+    const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
+    const int data_len = device->spec.size;
+    Uint8 *data;
+    void *udata = device->callbackspec.userdata;
+    SDL_AudioCallback callback = device->callbackspec.callback;
+
+    SDL_assert(device->iscapture);
+
+    /* The audio mixing is always a high priority thread */
+    SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
+
+    /* Perform any thread setup */
+    device->threadid = SDL_ThreadID();
+    current_audio.impl.ThreadInit(device);
+
+    /* Loop, filling the audio buffers */
+    while (!SDL_AtomicGet(&device->shutdown)) {
+        int still_need;
+        Uint8 *ptr;
+
+        current_audio.impl.BeginLoopIteration(device);
+
+        if (SDL_AtomicGet(&device->paused)) {
+            SDL_Delay(delay);  /* just so we don't cook the CPU. */
+            if (device->stream) {
+                SDL_AudioStreamClear(device->stream);
+            }
+            current_audio.impl.FlushCapture(device);  /* dump anything pending. */
+            continue;
+        }
+
+        /* Fill the current buffer with sound */
+        still_need = data_len;
+
+        /* Use the work_buffer to hold data read from the device. */
+        data = device->work_buffer;
+        SDL_assert(data != NULL);
+
+        ptr = data;
+
+        /* We still read from the device when "paused" to keep the state sane,
+           and block when there isn't data so this thread isn't eating CPU.
+           But we don't process it further or call the app's callback. */
+
+        if (!SDL_AtomicGet(&device->enabled)) {
+            SDL_Delay(delay);  /* try to keep callback firing at normal pace. */
+        } else {
+            while (still_need > 0) {
+                const int rc = current_audio.impl.CaptureFromDevice(device, ptr, still_need);
+                SDL_assert(rc <= still_need);  /* device should not overflow buffer. :) */
+                if (rc > 0) {
+                    still_need -= rc;
+                    ptr += rc;
+                } else {  /* uhoh, device failed for some reason! */
+                    SDL_OpenedAudioDeviceDisconnected(device);
+                    break;
+                }
+            }
+        }
+
+        if (still_need > 0) {
+            /* Keep any data we already read, silence the rest. */
+            SDL_memset(ptr, silence, still_need);
+        }
+
+        if (device->stream) {
+            /* if this fails...oh well. */
+            SDL_AudioStreamPut(device->stream, data, data_len);
+
+            while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->callbackspec.size)) {
+                const int got = SDL_AudioStreamGet(device->stream, device->work_buffer, device->callbackspec.size);
+                SDL_assert((got < 0) || (got == device->callbackspec.size));
+                if (got != device->callbackspec.size) {
+                    SDL_memset(device->work_buffer, device->spec.silence, device->callbackspec.size);
+                }
+
+                /* !!! FIXME: this should be LockDevice. */
+                SDL_LockMutex(device->mixer_lock);
+                if (!SDL_AtomicGet(&device->paused)) {
+                    callback(udata, device->work_buffer, device->callbackspec.size);
+                }
+                SDL_UnlockMutex(device->mixer_lock);
+            }
+        } else {  /* feeding user callback directly without streaming. */
+            /* !!! FIXME: this should be LockDevice. */
+            SDL_LockMutex(device->mixer_lock);
+            if (!SDL_AtomicGet(&device->paused)) {
+                callback(udata, data, device->callbackspec.size);
+            }
+            SDL_UnlockMutex(device->mixer_lock);
+        }
+    }
+
+    current_audio.impl.PrepareToClose(device);
+
+    current_audio.impl.FlushCapture(device);
+
+    current_audio.impl.ThreadDeinit(device);
+
+    return 0;
+}
+
+
+static SDL_AudioFormat
+SDL_ParseAudioFormat(const char *string)
+{
+#define CHECK_FMT_STRING(x) if (SDL_strcmp(string, #x) == 0) return AUDIO_##x
+    CHECK_FMT_STRING(U8);
+    CHECK_FMT_STRING(S8);
+    CHECK_FMT_STRING(U16LSB);
+    CHECK_FMT_STRING(S16LSB);
+    CHECK_FMT_STRING(U16MSB);
+    CHECK_FMT_STRING(S16MSB);
+    CHECK_FMT_STRING(U16SYS);
+    CHECK_FMT_STRING(S16SYS);
+    CHECK_FMT_STRING(U16);
+    CHECK_FMT_STRING(S16);
+    CHECK_FMT_STRING(S32LSB);
+    CHECK_FMT_STRING(S32MSB);
+    CHECK_FMT_STRING(S32SYS);
+    CHECK_FMT_STRING(S32);
+    CHECK_FMT_STRING(F32LSB);
+    CHECK_FMT_STRING(F32MSB);
+    CHECK_FMT_STRING(F32SYS);
+    CHECK_FMT_STRING(F32);
+#undef CHECK_FMT_STRING
+    return 0;
+}
+
+int
+SDL_GetNumAudioDrivers(void)
+{
+    return SDL_arraysize(bootstrap) - 1;
+}
+
+const char *
+SDL_GetAudioDriver(int index)
+{
+    if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
+        return bootstrap[index]->name;
+    }
+    return NULL;
+}
+
+int
+SDL_AudioInit(const char *driver_name)
+{
+    int i = 0;
+    int initialized = 0;
+    int tried_to_init = 0;
+
+    if (SDL_WasInit(SDL_INIT_AUDIO)) {
+        SDL_AudioQuit();        /* shutdown driver if already running. */
+    }
+
+    SDL_zero(current_audio);
+    SDL_zero(open_devices);
+
+    /* Select the proper audio driver */
+    if (driver_name == NULL) {
+        driver_name = SDL_getenv("SDL_AUDIODRIVER");
+    }
+
+    for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
+        /* make sure we should even try this driver before doing so... */
+        const AudioBootStrap *backend = bootstrap[i];
+        if ((driver_name && (SDL_strncasecmp(backend->name, driver_name, SDL_strlen(driver_name)) != 0)) ||
+            (!driver_name && backend->demand_only)) {
+            continue;
+        }
+
+        tried_to_init = 1;
+        SDL_zero(current_audio);
+        current_audio.name = backend->name;
+        current_audio.desc = backend->desc;
+        initialized = backend->init(&current_audio.impl);
+    }
+
+    if (!initialized) {
+        /* specific drivers will set the error message if they fail... */
+        if (!tried_to_init) {
+            if (driver_name) {
+                SDL_SetError("Audio target '%s' not available", driver_name);
+            } else {
+                SDL_SetError("No available audio device");
+            }
+        }
+
+        SDL_zero(current_audio);
+        return -1;            /* No driver was available, so fail. */
+    }
+
+    current_audio.detectionLock = SDL_CreateMutex();
+
+    finish_audio_entry_points_init();
+
+    /* Make sure we have a list of devices available at startup. */
+    current_audio.impl.DetectDevices();
+
+#ifdef HAVE_LIBSAMPLERATE_H
+    LoadLibSampleRate();
+#endif
+
+    return 0;
+}
+
+/*
+ * Get the current audio driver name
+ */
+const char *
+SDL_GetCurrentAudioDriver()
+{
+    return current_audio.name;
+}
+
+/* Clean out devices that we've removed but had to keep around for stability. */
+static void
+clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *removedFlag)
+{
+    SDL_AudioDeviceItem *item = *devices;
+    SDL_AudioDeviceItem *prev = NULL;
+    int total = 0;
+
+    while (item) {
+        SDL_AudioDeviceItem *next = item->next;
+        if (item->handle != NULL) {
+            total++;
+            prev = item;
+        } else {
+            if (prev) {
+                prev->next = next;
+            } else {
+                *devices = next;
+            }
+            SDL_free(item);
+        }
+        item = next;
+    }
+
+    *devCount = total;
+    *removedFlag = SDL_FALSE;
+}
+
+
+int
+SDL_GetNumAudioDevices(int iscapture)
+{
+    int retval = 0;
+
+    if (!SDL_WasInit(SDL_INIT_AUDIO)) {
+        return -1;
+    }
+
+    SDL_LockMutex(current_audio.detectionLock);
+    if (iscapture && current_audio.captureDevicesRemoved) {
+        clean_out_device_list(&current_audio.inputDevices, &current_audio.inputDeviceCount, &current_audio.captureDevicesRemoved);
+    }
+
+    if (!iscapture && current_audio.outputDevicesRemoved) {
+        clean_out_device_list(&current_audio.outputDevices, &current_audio.outputDeviceCount, &current_audio.outputDevicesRemoved);
+        current_audio.outputDevicesRemoved = SDL_FALSE;
+    }
+
+    retval = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
+    SDL_UnlockMutex(current_audio.detectionLock);
+
+    return retval;
+}
+
+
+const char *
+SDL_GetAudioDeviceName(int index, int iscapture)
+{
+    const char *retval = NULL;
+
+    if (!SDL_WasInit(SDL_INIT_AUDIO)) {
+        SDL_SetError("Audio subsystem is not initialized");
+        return NULL;
+    }
+
+    if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
+        SDL_SetError("No capture support");
+        return NULL;
+    }
+
+    if (index >= 0) {
+        SDL_AudioDeviceItem *item;
+        int i;
+
+        SDL_LockMutex(current_audio.detectionLock);
+        item = iscapture ? current_audio.inputDevices : current_audio.outputDevices;
+        i = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
+        if (index < i) {
+            for (i--; i > index; i--, item = item->next) {
+                SDL_assert(item != NULL);
+            }
+            SDL_assert(item != NULL);
+            retval = item->name;
+        }
+        SDL_UnlockMutex(current_audio.detectionLock);
+    }
+
+    if (retval == NULL) {
+        SDL_SetError("No such device");
+    }
+
+    return retval;
+}
+
+
+static void
+close_audio_device(SDL_AudioDevice * device)
+{
+    if (!device) {
+        return;
+    }
+
+    /* make sure the device is paused before we do anything else, so the
+       audio callback definitely won't fire again. */
+    current_audio.impl.LockDevice(device);
+    SDL_AtomicSet(&device->paused, 1);
+    SDL_AtomicSet(&device->shutdown, 1);
+    SDL_AtomicSet(&device->enabled, 0);
+    current_audio.impl.UnlockDevice(device);
+
+    if (device->thread != NULL) {
+        SDL_WaitThread(device->thread, NULL);
+    }
+    if (device->mixer_lock != NULL) {
+        SDL_DestroyMutex(device->mixer_lock);
+    }
+
+    SDL_free(device->work_buffer);
+    SDL_FreeAudioStream(device->stream);
+
+    if (device->id > 0) {
+        SDL_AudioDevice *opendev = open_devices[device->id - 1];
+        SDL_assert((opendev == device) || (opendev == NULL));
+        if (opendev == device) {
+            open_devices[device->id - 1] = NULL;
+        }
+    }
+
+    if (device->hidden != NULL) {
+        current_audio.impl.CloseDevice(device);
+    }
+
+    SDL_FreeDataQueue(device->buffer_queue);
+
+    SDL_free(device);
+}
+
+
+/*
+ * Sanity check desired AudioSpec for SDL_OpenAudio() in (orig).
+ *  Fills in a sanitized copy in (prepared).
+ *  Returns non-zero if okay, zero on fatal parameters in (orig).
+ */
+static int
+prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
+{
+    SDL_memcpy(prepared, orig, sizeof(SDL_AudioSpec));
+
+    if (orig->freq == 0) {
+        const char *env = SDL_getenv("SDL_AUDIO_FREQUENCY");
+        if ((!env) || ((prepared->freq = SDL_atoi(env)) == 0)) {
+            prepared->freq = 22050;     /* a reasonable default */
+        }
+    }
+
+    if (orig->format == 0) {
+        const char *env = SDL_getenv("SDL_AUDIO_FORMAT");
+        if ((!env) || ((prepared->format = SDL_ParseAudioFormat(env)) == 0)) {
+            prepared->format = AUDIO_S16;       /* a reasonable default */
+        }
+    }
+
+    switch (orig->channels) {
+    case 0:{
+            const char *env = SDL_getenv("SDL_AUDIO_CHANNELS");
+            if ((!env) || ((prepared->channels = (Uint8) SDL_atoi(env)) == 0)) {
+                prepared->channels = 2; /* a reasonable default */
+            }
+            break;
+        }
+    case 1:                    /* Mono */
+    case 2:                    /* Stereo */
+    case 4:                    /* surround */
+    case 6:                    /* surround with center and lfe */
+        break;
+    default:
+        SDL_SetError("Unsupported number of audio channels.");
+        return 0;
+    }
+
+    if (orig->samples == 0) {
+        const char *env = SDL_getenv("SDL_AUDIO_SAMPLES");
+        if ((!env) || ((prepared->samples = (Uint16) SDL_atoi(env)) == 0)) {
+            /* Pick a default of ~46 ms at desired frequency */
+            /* !!! FIXME: remove this when the non-Po2 resampling is in. */
+            const int samples = (prepared->freq / 1000) * 46;
+            int power2 = 1;
+            while (power2 < samples) {
+                power2 *= 2;
+            }
+            prepared->samples = power2;
+        }
+    }
+
+    /* Calculate the silence and size of the audio specification */
+    SDL_CalculateAudioSpec(prepared);
+
+    return 1;
+}
+
+static SDL_AudioDeviceID
+open_audio_device(const char *devname, int iscapture,
+                  const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
+                  int allowed_changes, int min_id)
+{
+    const SDL_bool is_internal_thread = (desired->callback == NULL);
+    SDL_AudioDeviceID id = 0;
+    SDL_AudioSpec _obtained;
+    SDL_AudioDevice *device;
+    SDL_bool build_stream;
+    void *handle = NULL;
+    int i = 0;
+
+    if (!SDL_WasInit(SDL_INIT_AUDIO)) {
+        SDL_SetError("Audio subsystem is not initialized");
+        return 0;
+    }
+
+    if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
+        SDL_SetError("No capture support");
+        return 0;
+    }
+
+    /* !!! FIXME: there is a race condition here if two devices open from two threads at once. */
+    /* Find an available device ID... */
+    for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
+        if (open_devices[id] == NULL) {
+            break;
+        }
+    }
+
+    if (id == SDL_arraysize(open_devices)) {
+        SDL_SetError("Too many open audio devices");
+        return 0;
+    }
+
+    if (!obtained) {
+        obtained = &_obtained;
+    }
+    if (!prepare_audiospec(desired, obtained)) {
+        return 0;
+    }
+
+    /* If app doesn't care about a specific device, let the user override. */
+    if (devname == NULL) {
+        devname = SDL_getenv("SDL_AUDIO_DEVICE_NAME");
+    }
+
+    /*
+     * Catch device names at the high level for the simple case...
+     * This lets us have a basic "device enumeration" for systems that
+     *  don't have multiple devices, but makes sure the device name is
+     *  always NULL when it hits the low level.
+     *
+     * Also make sure that the simple case prevents multiple simultaneous
+     *  opens of the default system device.
+     */
+
+    if ((iscapture) && (current_audio.impl.OnlyHasDefaultCaptureDevice)) {
+        if ((devname) && (SDL_strcmp(devname, DEFAULT_INPUT_DEVNAME) != 0)) {
+            SDL_SetError("No such device");
+            return 0;
+        }
+        devname = NULL;
+
+        for (i = 0; i < SDL_arraysize(open_devices); i++) {
+            if ((open_devices[i]) && (open_devices[i]->iscapture)) {
+                SDL_SetError("Audio device already open");
+                return 0;
+            }
+        }
+    } else if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
+        if ((devname) && (SDL_strcmp(devname, DEFAULT_OUTPUT_DEVNAME) != 0)) {
+            SDL_SetError("No such device");
+            return 0;
+        }
+        devname = NULL;
+
+        for (i = 0; i < SDL_arraysize(open_devices); i++) {
+            if ((open_devices[i]) && (!open_devices[i]->iscapture)) {
+                SDL_SetError("Audio device already open");
+                return 0;
+            }
+        }
+    } else if (devname != NULL) {
+        /* if the app specifies an exact string, we can pass the backend
+           an actual device handle thingey, which saves them the effort of
+           figuring out what device this was (such as, reenumerating
+           everything again to find the matching human-readable name).
+           It might still need to open a device based on the string for,
+           say, a network audio server, but this optimizes some cases. */
+        SDL_AudioDeviceItem *item;
+        SDL_LockMutex(current_audio.detectionLock);
+        for (item = iscapture ? current_audio.inputDevices : current_audio.outputDevices; item; item = item->next) {
+            if ((item->handle != NULL) && (SDL_strcmp(item->name, devname) == 0)) {
+                handle = item->handle;
+                break;
+            }
+        }
+        SDL_UnlockMutex(current_audio.detectionLock);
+    }
+
+    if (!current_audio.impl.AllowsArbitraryDeviceNames) {
+        /* has to be in our device list, or the default device. */
+        if ((handle == NULL) && (devname != NULL)) {
+            SDL_SetError("No such device.");
+            return 0;
+        }
+    }
+
+    device = (SDL_AudioDevice *) SDL_calloc(1, sizeof (SDL_AudioDevice));
+    if (device == NULL) {
+        SDL_OutOfMemory();
+        return 0;
+    }
+    device->id = id + 1;
+    device->spec = *obtained;
+    device->iscapture = iscapture ? SDL_TRUE : SDL_FALSE;
+    device->handle = handle;
+
+    SDL_AtomicSet(&device->shutdown, 0);  /* just in case. */
+    SDL_AtomicSet(&device->paused, 1);
+    SDL_AtomicSet(&device->enabled, 1);
+
+    /* Create a mutex for locking the sound buffers */
+    if (!current_audio.impl.SkipMixerLock) {
+        device->mixer_lock = SDL_CreateMutex();
+        if (device->mixer_lock == NULL) {
+            close_audio_device(device);
+            SDL_SetError("Couldn't create mixer lock");
+            return 0;
+        }
+    }
+
+    if (current_audio.impl.OpenDevice(device, handle, devname, iscapture) < 0) {
+        close_audio_device(device);
+        return 0;
+    }
+
+    /* if your target really doesn't need it, set it to 0x1 or something. */
+    /* otherwise, close_audio_device() won't call impl.CloseDevice(). */
+    SDL_assert(device->hidden != NULL);
+
+    /* See if we need to do any conversion */
+    build_stream = SDL_FALSE;
+    if (obtained->freq != device->spec.freq) {
+        if (allowed_changes & SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) {
+            obtained->freq = device->spec.freq;
+        } else {
+            build_stream = SDL_TRUE;
+        }
+    }
+    if (obtained->format != device->spec.format) {
+        if (allowed_changes & SDL_AUDIO_ALLOW_FORMAT_CHANGE) {
+            obtained->format = device->spec.format;
+        } else {
+            build_stream = SDL_TRUE;
+        }
+    }
+    if (obtained->channels != device->spec.channels) {
+        if (allowed_changes & SDL_AUDIO_ALLOW_CHANNELS_CHANGE) {
+            obtained->channels = device->spec.channels;
+        } else {
+            build_stream = SDL_TRUE;
+        }
+    }
+
+    /* !!! FIXME in 2.1: add SDL_AUDIO_ALLOW_SAMPLES_CHANGE flag?
+       As of 2.0.6, we will build a stream to buffer the difference between
+       what the app wants to feed and the device wants to eat, so everyone
+       gets their way. In prior releases, SDL would force the callback to
+       feed at the rate the device requested, adjusted for resampling.
+     */
+    if (device->spec.samples != obtained->samples) {
+        build_stream = SDL_TRUE;
+    }
+
+    SDL_CalculateAudioSpec(obtained);  /* recalc after possible changes. */
+
+    device->callbackspec = *obtained;
+
+    if (build_stream) {
+        if (iscapture) {
+            device->stream = SDL_NewAudioStream(device->spec.format,
+                                  device->spec.channels, device->spec.freq,
+                                  obtained->format, obtained->channels, obtained->freq);
+        } else {
+            device->stream = SDL_NewAudioStream(obtained->format, obtained->channels,
+                                  obtained->freq, device->spec.format,
+                                  device->spec.channels, device->spec.freq);
+        }
+
+        if (!device->stream) {
+            close_audio_device(device);
+            return 0;
+        }
+    }
+
+    if (device->spec.callback == NULL) {  /* use buffer queueing? */
+        /* pool a few packets to start. Enough for two callbacks. */
+        device->buffer_queue = SDL_NewDataQueue(SDL_AUDIOBUFFERQUEUE_PACKETLEN, obtained->size * 2);
+        if (!device->buffer_queue) {
+            close_audio_device(device);
+            SDL_SetError("Couldn't create audio buffer queue");
+            return 0;
+        }
+        device->callbackspec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback;
+        device->callbackspec.userdata = device;
+    }
+
+    /* Allocate a scratch audio buffer */
+    device->work_buffer_len = build_stream ? device->callbackspec.size : 0;
+    if (device->spec.size > device->work_buffer_len) {
+        device->work_buffer_len = device->spec.size;
+    }
+    SDL_assert(device->work_buffer_len > 0);
+
+    device->work_buffer = (Uint8 *) SDL_malloc(device->work_buffer_len);
+    if (device->work_buffer == NULL) {
+        close_audio_device(device);
+        SDL_OutOfMemory();
+        return 0;
+    }
+
+    open_devices[id] = device;  /* add it to our list of open devices. */
+
+    /* Start the audio thread if necessary */
+    if (!current_audio.impl.ProvidesOwnCallbackThread) {
+        /* Start the audio thread */
+        /* !!! FIXME: we don't force the audio thread stack size here if it calls into user code, but maybe we should? */
+        /* buffer queueing callback only needs a few bytes, so make the stack tiny. */
+        const size_t stacksize = is_internal_thread ? 64 * 1024 : 0;
+        char threadname[64];
+
+        SDL_snprintf(threadname, sizeof (threadname), "SDLAudio%c%d", (iscapture) ? 'C' : 'P', (int) device->id);
+        device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, threadname, stacksize, device);
+
+        if (device->thread == NULL) {
+            close_audio_device(device);
+            SDL_SetError("Couldn't create audio thread");
+            return 0;
+        }
+    }
+
+    return device->id;
+}
+
+
+int
+SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
+{
+    SDL_AudioDeviceID id = 0;
+
+    /* Start up the audio driver, if necessary. This is legacy behaviour! */
+    if (!SDL_WasInit(SDL_INIT_AUDIO)) {
+        if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
+            return -1;
+        }
+    }
+
+    /* SDL_OpenAudio() is legacy and can only act on Device ID #1. */
+    if (open_devices[0] != NULL) {
+        SDL_SetError("Audio device is already opened");
+        return -1;
+    }
+
+    if (obtained) {
+        id = open_audio_device(NULL, 0, desired, obtained,
+                               SDL_AUDIO_ALLOW_ANY_CHANGE, 1);
+    } else {
+        SDL_AudioSpec _obtained;
+        SDL_zero(_obtained);
+        id = open_audio_device(NULL, 0, desired, &_obtained, 0, 1);
+        /* On successful open, copy calculated values into 'desired'. */
+        if (id > 0) {
+            desired->size = _obtained.size;
+            desired->silence = _obtained.silence;
+        }
+    }
+
+    SDL_assert((id == 0) || (id == 1));
+    return (id == 0) ? -1 : 0;
+}
+
+SDL_AudioDeviceID
+SDL_OpenAudioDevice(const char *device, int iscapture,
+                    const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
+                    int allowed_changes)
+{
+    return open_audio_device(device, iscapture, desired, obtained,
+                             allowed_changes, 2);
+}
+
+SDL_AudioStatus
+SDL_GetAudioDeviceStatus(SDL_AudioDeviceID devid)
+{
+    SDL_AudioDevice *device = get_audio_device(devid);
+    SDL_AudioStatus status = SDL_AUDIO_STOPPED;
+    if (device && SDL_AtomicGet(&device->enabled)) {
+        if (SDL_AtomicGet(&device->paused)) {
+            status = SDL_AUDIO_PAUSED;
+        } else {
+            status = SDL_AUDIO_PLAYING;
+        }
+    }
+    return status;
+}
+
+
+SDL_AudioStatus
+SDL_GetAudioStatus(void)
+{
+    return SDL_GetAudioDeviceStatus(1);
+}
+
+void
+SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
+{
+    SDL_AudioDevice *device = get_audio_device(devid);
+    if (device) {
+        current_audio.impl.LockDevice(device);
+        SDL_AtomicSet(&device->paused, pause_on ? 1 : 0);
+        current_audio.impl.UnlockDevice(device);
+    }
+}
+
+void
+SDL_PauseAudio(int pause_on)
+{
+    SDL_PauseAudioDevice(1, pause_on);
+}
+
+
+void
+SDL_LockAudioDevice(SDL_AudioDeviceID devid)
+{
+    /* Obtain a lock on the mixing buffers */
+    SDL_AudioDevice *device = get_audio_device(devid);
+    if (device) {
+        current_audio.impl.LockDevice(device);
+    }
+}
+
+void
+SDL_LockAudio(void)
+{
+    SDL_LockAudioDevice(1);
+}
+
+void
+SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
+{
+    /* Obtain a lock on the mixing buffers */
+    SDL_AudioDevice *device = get_audio_device(devid);
+    if (device) {
+        current_audio.impl.UnlockDevice(device);
+    }
+}
+
+void
+SDL_UnlockAudio(void)
+{
+    SDL_UnlockAudioDevice(1);
+}
+
+void
+SDL_CloseAudioDevice(SDL_AudioDeviceID devid)
+{
+    close_audio_device(get_audio_device(devid));
+}
+
+void
+SDL_CloseAudio(void)
+{
+    SDL_CloseAudioDevice(1);
+}
+
+void
+SDL_AudioQuit(void)
+{
+    SDL_AudioDeviceID i;
+
+    if (!current_audio.name) {  /* not initialized?! */
+        return;
+    }
+
+    for (i = 0; i < SDL_arraysize(open_devices); i++) {
+        close_audio_device(open_devices[i]);
+    }
+
+    free_device_list(&current_audio.outputDevices, &current_audio.outputDeviceCount);
+    free_device_list(&current_audio.inputDevices, &current_audio.inputDeviceCount);
+
+    /* Free the driver data */
+    current_audio.impl.Deinitialize();
+
+    SDL_DestroyMutex(current_audio.detectionLock);
+
+    SDL_zero(current_audio);
+    SDL_zero(open_devices);
+
+#ifdef HAVE_LIBSAMPLERATE_H
+    UnloadLibSampleRate();
+#endif
+
+    SDL_FreeResampleFilter();
+}
+
+#define NUM_FORMATS 10
+static int format_idx;
+static int format_idx_sub;
+static SDL_AudioFormat format_list[NUM_FORMATS][NUM_FORMATS] = {
+    {AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
+     AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB},
+    {AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
+     AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB},
+    {AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S32LSB,
+     AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S32MSB,
+     AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_S32LSB,
+     AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_S32MSB,
+     AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S16LSB,
+     AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S16MSB,
+     AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_S16LSB,
+     AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8},
+    {AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_S16MSB,
+     AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8},
+};
+
+SDL_AudioFormat
+SDL_FirstAudioFormat(SDL_AudioFormat format)
+{
+    for (format_idx = 0; format_idx < NUM_FORMATS; ++format_idx) {
+        if (format_list[format_idx][0] == format) {
+            break;
+        }
+    }
+    format_idx_sub = 0;
+    return SDL_NextAudioFormat();
+}
+
+SDL_AudioFormat
+SDL_NextAudioFormat(void)
+{
+    if ((format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS)) {
+        return 0;
+    }
+    return format_list[format_idx][format_idx_sub++];
+}
+
+void
+SDL_CalculateAudioSpec(SDL_AudioSpec * spec)
+{
+    switch (spec->format) {
+    case AUDIO_U8:
+        spec->silence = 0x80;
+        break;
+    default:
+        spec->silence = 0x00;
+        break;
+    }
+    spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
+    spec->size *= spec->channels;
+    spec->size *= spec->samples;
+}
+
+
+/*
+ * Moved here from SDL_mixer.c, since it relies on internals of an opened
+ *  audio device (and is deprecated, by the way!).
+ */
+void
+SDL_MixAudio(Uint8 * dst, const Uint8 * src, Uint32 len, int volume)
+{
+    /* Mix the user-level audio format */
+    SDL_AudioDevice *device = get_audio_device(1);
+    if (device != NULL) {
+        SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
+    }
+}
+
+/* vi: set ts=4 sw=4 expandtab: */