-Reverted some changes to make the code build again on Tiger.

-Removed x86_64 from Deployment configuration.
-macosx: Use SDL for sound plugin, removed Carbon backend.
-(MaddTheSane)Fixed memory leaks (Patch #8427).

git-svn-id: https://pcsxr.svn.codeplex.com/svn/pcsxr@63548 e17a0e51-4ae3-4d35-97c3-1a29b211df97

1 files changed, 1121 insertions, 0 deletions

diff --git a/macosx/plugins/Common/SDL/src/audio/SDL_audio.c b/macosx/plugins/Common/SDL/src/audio/SDL_audio.c
new file mode 100644
index 00000000..bd0a5430
--- /dev/null
+++ b/macosx/plugins/Common/SDL/src/audio/SDL_audio.c
@@ -0,0 +1,1121 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2010 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to a raw mixing buffer */
+
+#include "SDL.h"
+#include "SDL_audio.h"
+#include "SDL_audio_c.h"
+#include "SDL_audiomem.h"
+#include "SDL_sysaudio.h"
+
+#define _THIS SDL_AudioDevice *_this
+
+static SDL_AudioDriver current_audio;
+static SDL_AudioDevice *open_devices[16];
+
+/* !!! FIXME: These are wordy and unlocalized... */
+#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
+#define DEFAULT_INPUT_DEVNAME "System audio capture device"
+
+
+/*
+ * Not all of these will be compiled and linked in, but it's convenient
+ *  to have a complete list here and saves yet-another block of #ifdefs...
+ *  Please see bootstrap[], below, for the actual #ifdef mess.
+ */
+
+extern AudioBootStrap COREAUDIO_bootstrap;
+
+/* Available audio drivers */
+static const AudioBootStrap *const bootstrap[] = {
+    &COREAUDIO_bootstrap, NULL
+};
+
+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 int
+SDL_AudioDetectDevices_Default(int iscapture)
+{
+    return -1;
+}
+
+static void
+SDL_AudioThreadInit_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioWaitDevice_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioPlayDevice_Default(_THIS)
+{                               /* no-op. */
+}
+
+static Uint8 *
+SDL_AudioGetDeviceBuf_Default(_THIS)
+{
+    return NULL;
+}
+
+static void
+SDL_AudioWaitDone_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioCloseDevice_Default(_THIS)
+{                               /* no-op. */
+}
+
+static void
+SDL_AudioDeinitialize_Default(void)
+{                               /* no-op. */
+}
+
+static int
+SDL_AudioOpenDevice_Default(_THIS, const char *devname, int iscapture)
+{
+    return 0;
+}
+
+static const char *
+SDL_AudioGetDeviceName_Default(int index, int iscapture)
+{
+    SDL_SetError("No such device");
+    return NULL;
+}
+
+static void
+SDL_AudioLockDevice_Default(SDL_AudioDevice * device)
+{
+    if (device->thread && (SDL_ThreadID() == device->threadid)) {
+        return;
+    }
+    SDL_mutexP(device->mixer_lock);
+}
+
+static void
+SDL_AudioUnlockDevice_Default(SDL_AudioDevice * device)
+{
+    if (device->thread && (SDL_ThreadID() == device->threadid)) {
+        return;
+    }
+    SDL_mutexV(device->mixer_lock);
+}
+
+
+static void
+finalize_audio_entry_points(void)
+{
+    /*
+     * Fill in stub functions for unused driver entry points. This lets us
+     *  blindly call them without having to check for validity first.
+     */
+
+#define FILL_STUB(x) \
+        if (current_audio.impl.x == NULL) { \
+            current_audio.impl.x = SDL_Audio##x##_Default; \
+        }
+    FILL_STUB(DetectDevices);
+    FILL_STUB(GetDeviceName);
+    FILL_STUB(OpenDevice);
+    FILL_STUB(ThreadInit);
+    FILL_STUB(WaitDevice);
+    FILL_STUB(PlayDevice);
+    FILL_STUB(GetDeviceBuf);
+    FILL_STUB(WaitDone);
+    FILL_STUB(CloseDevice);
+    FILL_STUB(LockDevice);
+    FILL_STUB(UnlockDevice);
+    FILL_STUB(Deinitialize);
+#undef FILL_STUB
+}
+
+/* Streaming functions (for when the input and output buffer sizes are different) */
+/* Write [length] bytes from buf into the streamer */
+static void
+SDL_StreamWrite(SDL_AudioStreamer * stream, Uint8 * buf, int length)
+{
+    int i;
+
+    for (i = 0; i < length; ++i) {
+        stream->buffer[stream->write_pos] = buf[i];
+        ++stream->write_pos;
+    }
+}
+
+/* Read [length] bytes out of the streamer into buf */
+static void
+SDL_StreamRead(SDL_AudioStreamer * stream, Uint8 * buf, int length)
+{
+    int i;
+
+    for (i = 0; i < length; ++i) {
+        buf[i] = stream->buffer[stream->read_pos];
+        ++stream->read_pos;
+    }
+}
+
+static int
+SDL_StreamLength(SDL_AudioStreamer * stream)
+{
+    return (stream->write_pos - stream->read_pos) % stream->max_len;
+}
+
+/* Initialize the stream by allocating the buffer and setting the read/write heads to the beginning */
+#if 0
+static int
+SDL_StreamInit(SDL_AudioStreamer * stream, int max_len, Uint8 silence)
+{
+    /* First try to allocate the buffer */
+    stream->buffer = (Uint8 *) SDL_malloc(max_len);
+    if (stream->buffer == NULL) {
+        return -1;
+    }
+
+    stream->max_len = max_len;
+    stream->read_pos = 0;
+    stream->write_pos = 0;
+
+    /* Zero out the buffer */
+    SDL_memset(stream->buffer, silence, max_len);
+
+    return 0;
+}
+#endif
+
+/* Deinitialize the stream simply by freeing the buffer */
+static void
+SDL_StreamDeinit(SDL_AudioStreamer * stream)
+{
+    if (stream->buffer != NULL) {
+        SDL_free(stream->buffer);
+    }
+}
+
+/* The general mixing thread function */
+int SDLCALL
+SDL_RunAudio(void *devicep)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
+    Uint8 *stream;
+    int stream_len;
+    void *udata;
+    void (SDLCALL * fill) (void *userdata, Uint8 * stream, int len);
+    int silence;
+    Uint32 delay;
+
+    /* For streaming when the buffer sizes don't match up */
+    Uint8 *istream;
+    int istream_len = 0;
+
+    /* Perform any thread setup */
+    device->threadid = SDL_ThreadID();
+    current_audio.impl.ThreadInit(device);
+
+    /* Set up the mixing function */
+    fill = device->spec.callback;
+    udata = device->spec.userdata;
+
+    /* By default do not stream */
+    device->use_streamer = 0;
+
+    if (device->convert.needed) {
+        if (device->convert.src_format == AUDIO_U8) {
+            silence = 0x80;
+        } else {
+            silence = 0;
+        }
+
+#if 0                           /* !!! FIXME: I took len_div out of the structure. Use rate_incr instead? */
+        /* If the result of the conversion alters the length, i.e. resampling is being used, use the streamer */
+        if (device->convert.len_mult != 1 || device->convert.len_div != 1) {
+            /* The streamer's maximum length should be twice whichever is larger: spec.size or len_cvt */
+            stream_max_len = 2 * device->spec.size;
+            if (device->convert.len_mult > device->convert.len_div) {
+                stream_max_len *= device->convert.len_mult;
+                stream_max_len /= device->convert.len_div;
+            }
+            if (SDL_StreamInit(&device->streamer, stream_max_len, silence) <
+                0)
+                return -1;
+            device->use_streamer = 1;
+
+            /* istream_len should be the length of what we grab from the callback and feed to conversion,
+               so that we get close to spec_size. I.e. we want device.spec_size = istream_len * u / d
+             */
+            istream_len =
+                device->spec.size * device->convert.len_div /
+                device->convert.len_mult;
+        }
+#endif
+
+        /* stream_len = device->convert.len; */
+        stream_len = device->spec.size;
+    } else {
+        silence = device->spec.silence;
+        stream_len = device->spec.size;
+    }
+
+    /* Calculate the delay while paused */
+    delay = ((device->spec.samples * 1000) / device->spec.freq);
+
+    /* Determine if the streamer is necessary here */
+    if (device->use_streamer == 1) {
+        /* This code is almost the same as the old code. The difference is, instead of reading
+           directly from the callback into "stream", then converting and sending the audio off,
+           we go: callback -> "istream" -> (conversion) -> streamer -> stream -> device.
+           However, reading and writing with streamer are done separately:
+           - We only call the callback and write to the streamer when the streamer does not
+           contain enough samples to output to the device.
+           - We only read from the streamer and tell the device to play when the streamer
+           does have enough samples to output.
+           This allows us to perform resampling in the conversion step, where the output of the
+           resampling process can be any number. We will have to see what a good size for the
+           stream's maximum length is, but I suspect 2*max(len_cvt, stream_len) is a good figure.
+         */
+        while (device->enabled) {
+
+            if (device->paused) {
+                SDL_Delay(delay);
+                continue;
+            }
+
+            /* Only read in audio if the streamer doesn't have enough already (if it does not have enough samples to output) */
+            if (SDL_StreamLength(&device->streamer) < stream_len) {
+                /* Set up istream */
+                if (device->convert.needed) {
+                    if (device->convert.buf) {
+                        istream = device->convert.buf;
+                    } else {
+                        continue;
+                    }
+                } else {
+/* FIXME: Ryan, this is probably wrong.  I imagine we don't want to get
+ * a device buffer both here and below in the stream output.
+ */
+                    istream = current_audio.impl.GetDeviceBuf(device);
+                    if (istream == NULL) {
+                        istream = device->fake_stream;
+                    }
+                }
+
+                /* Read from the callback into the _input_ stream */
+                SDL_mutexP(device->mixer_lock);
+                (*fill) (udata, istream, istream_len);
+                SDL_mutexV(device->mixer_lock);
+
+                /* Convert the audio if necessary and write to the streamer */
+                if (device->convert.needed) {
+                    SDL_ConvertAudio(&device->convert);
+                    if (istream == NULL) {
+                        istream = device->fake_stream;
+                    }
+                    /*SDL_memcpy(istream, device->convert.buf, device->convert.len_cvt); */
+                    SDL_StreamWrite(&device->streamer, device->convert.buf,
+                                    device->convert.len_cvt);
+                } else {
+                    SDL_StreamWrite(&device->streamer, istream, istream_len);
+                }
+            }
+
+            /* Only output audio if the streamer has enough to output */
+            if (SDL_StreamLength(&device->streamer) >= stream_len) {
+                /* Set up the output stream */
+                if (device->convert.needed) {
+                    if (device->convert.buf) {
+                        stream = device->convert.buf;
+                    } else {
+                        continue;
+                    }
+                } else {
+                    stream = current_audio.impl.GetDeviceBuf(device);
+                    if (stream == NULL) {
+                        stream = device->fake_stream;
+                    }
+                }
+
+                /* Now read from the streamer */
+                SDL_StreamRead(&device->streamer, stream, stream_len);
+
+                /* Ready current buffer for play and change current buffer */
+                if (stream != device->fake_stream) {
+                    current_audio.impl.PlayDevice(device);
+                    /* Wait for an audio buffer to become available */
+                    current_audio.impl.WaitDevice(device);
+                } else {
+                    SDL_Delay(delay);
+                }
+            }
+
+        }
+    } else {
+        /* Otherwise, do not use the streamer. This is the old code. */
+
+        /* Loop, filling the audio buffers */
+        while (device->enabled) {
+
+            if (device->paused) {
+                SDL_Delay(delay);
+                continue;
+            }
+
+            /* Fill the current buffer with sound */
+            if (device->convert.needed) {
+                if (device->convert.buf) {
+                    stream = device->convert.buf;
+                } else {
+                    continue;
+                }
+            } else {
+                stream = current_audio.impl.GetDeviceBuf(device);
+                if (stream == NULL) {
+                    stream = device->fake_stream;
+                }
+            }
+
+            SDL_mutexP(device->mixer_lock);
+            (*fill) (udata, stream, stream_len);
+            SDL_mutexV(device->mixer_lock);
+
+            /* Convert the audio if necessary */
+            if (device->convert.needed) {
+                SDL_ConvertAudio(&device->convert);
+                stream = current_audio.impl.GetDeviceBuf(device);
+                if (stream == NULL) {
+                    stream = device->fake_stream;
+                }
+                SDL_memcpy(stream, device->convert.buf,
+                           device->convert.len_cvt);
+            }
+
+            /* Ready current buffer for play and change current buffer */
+            if (stream != device->fake_stream) {
+                current_audio.impl.PlayDevice(device);
+                /* Wait for an audio buffer to become available */
+                current_audio.impl.WaitDevice(device);
+            } else {
+                SDL_Delay(delay);
+            }
+        }
+    }
+
+    /* Wait for the audio to drain.. */
+    current_audio.impl.WaitDone(device);
+
+    /* If necessary, deinit the streamer */
+    if (device->use_streamer == 1)
+        SDL_StreamDeinit(&device->streamer);
+
+    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_memset(&current_audio, '\0', sizeof(current_audio));
+    SDL_memset(open_devices, '\0', sizeof(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_strcasecmp(backend->name, driver_name))) ||
+            ((!driver_name) && (backend->demand_only))) {
+            continue;
+        }
+
+        tried_to_init = 1;
+        SDL_memset(&current_audio, 0, sizeof(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_memset(&current_audio, 0, sizeof(current_audio));
+        return (-1);            /* No driver was available, so fail. */
+    }
+
+    finalize_audio_entry_points();
+
+    return (0);
+}
+
+/*
+ * Get the current audio driver name
+ */
+const char *
+SDL_GetCurrentAudioDriver()
+{
+    return current_audio.name;
+}
+
+
+int
+SDL_GetNumAudioDevices(int iscapture)
+{
+    if (!SDL_WasInit(SDL_INIT_AUDIO)) {
+        return -1;
+    }
+    if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
+        return 0;
+    }
+
+    if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
+        return 1;
+    }
+
+    if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
+        return 1;
+    }
+
+    return current_audio.impl.DetectDevices(iscapture);
+}
+
+
+const char *
+SDL_GetAudioDeviceName(int index, int iscapture)
+{
+    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_SetError("No such device");
+        return NULL;
+    }
+
+    if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
+        return DEFAULT_INPUT_DEVNAME;
+    }
+
+    if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
+        return DEFAULT_OUTPUT_DEVNAME;
+    }
+
+    return current_audio.impl.GetDeviceName(index, iscapture);
+}
+
+
+static void
+close_audio_device(SDL_AudioDevice * device)
+{
+    device->enabled = 0;
+    if (device->thread != NULL) {
+        SDL_WaitThread(device->thread, NULL);
+    }
+    if (device->mixer_lock != NULL) {
+        SDL_DestroyMutex(device->mixer_lock);
+    }
+    if (device->fake_stream != NULL) {
+        SDL_FreeAudioMem(device->fake_stream);
+    }
+    if (device->convert.needed) {
+        SDL_FreeAudioMem(device->convert.buf);
+    }
+    if (device->opened) {
+        current_audio.impl.CloseDevice(device);
+        device->opened = 0;
+    }
+    SDL_FreeAudioMem(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->callback == NULL) {
+        SDL_SetError("SDL_OpenAudio() passed a NULL callback");
+        return 0;
+    }
+
+    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)
+{
+    SDL_AudioDeviceID id = 0;
+    SDL_AudioSpec _obtained;
+    SDL_AudioDevice *device;
+    SDL_bool build_cvt;
+    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;
+    }
+
+    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.OnlyHasDefaultInputDevice)) {
+        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;
+            }
+        }
+    }
+
+    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;
+            }
+        }
+    }
+
+    device = (SDL_AudioDevice *) SDL_AllocAudioMem(sizeof(SDL_AudioDevice));
+    if (device == NULL) {
+        SDL_OutOfMemory();
+        return 0;
+    }
+    SDL_memset(device, '\0', sizeof(SDL_AudioDevice));
+    device->spec = *obtained;
+    device->enabled = 1;
+    device->paused = 1;
+    device->iscapture = iscapture;
+
+    /* Create a semaphore 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, devname, iscapture)) {
+        close_audio_device(device);
+        return 0;
+    }
+    device->opened = 1;
+
+    /* Allocate a fake audio memory buffer */
+    device->fake_stream = (Uint8 *)SDL_AllocAudioMem(device->spec.size);
+    if (device->fake_stream == NULL) {
+        close_audio_device(device);
+        SDL_OutOfMemory();
+        return 0;
+    }
+
+    /* If the audio driver changes the buffer size, accept it */
+    if (device->spec.samples != obtained->samples) {
+        obtained->samples = device->spec.samples;
+        SDL_CalculateAudioSpec(obtained);
+    }
+
+    /* See if we need to do any conversion */
+    build_cvt = SDL_FALSE;
+    if (obtained->freq != device->spec.freq) {
+        if (allowed_changes & SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) {
+            obtained->freq = device->spec.freq;
+        } else {
+            build_cvt = SDL_TRUE;
+        }
+    }
+    if (obtained->format != device->spec.format) {
+        if (allowed_changes & SDL_AUDIO_ALLOW_FORMAT_CHANGE) {
+            obtained->format = device->spec.format;
+        } else {
+            build_cvt = SDL_TRUE;
+        }
+    }
+    if (obtained->channels != device->spec.channels) {
+        if (allowed_changes & SDL_AUDIO_ALLOW_CHANNELS_CHANGE) {
+            obtained->channels = device->spec.channels;
+        } else {
+            build_cvt = SDL_TRUE;
+        }
+    }
+    if (build_cvt) {
+        /* Build an audio conversion block */
+        if (SDL_BuildAudioCVT(&device->convert,
+                              obtained->format, obtained->channels,
+                              obtained->freq,
+                              device->spec.format, device->spec.channels,
+                              device->spec.freq) < 0) {
+            close_audio_device(device);
+            return 0;
+        }
+        if (device->convert.needed) {
+            device->convert.len = (int) (((double) obtained->size) /
+                                         device->convert.len_ratio);
+
+            device->convert.buf =
+                (Uint8 *) SDL_AllocAudioMem(device->convert.len *
+                                            device->convert.len_mult);
+            if (device->convert.buf == NULL) {
+                close_audio_device(device);
+                SDL_OutOfMemory();
+                return 0;
+            }
+        }
+    }
+
+    /* Find an available device ID and store the structure... */
+    for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
+        if (open_devices[id] == NULL) {
+            open_devices[id] = device;
+            break;
+        }
+    }
+
+    if (id == SDL_arraysize(open_devices)) {
+        SDL_SetError("Too many open audio devices");
+        close_audio_device(device);
+        return 0;
+    }
+
+    /* Start the audio thread if necessary */
+    if (!current_audio.impl.ProvidesOwnCallbackThread) {
+        /* Start the audio thread */
+/* !!! FIXME: this is nasty. */
+#if (defined(__WIN32__) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC)
+#undef SDL_CreateThread
+        device->thread = SDL_CreateThread(SDL_RunAudio, device, NULL, NULL);
+#else
+        device->thread = SDL_CreateThread(SDL_RunAudio, device);
+#endif
+        if (device->thread == NULL) {
+            SDL_CloseAudioDevice(id + 1);
+            SDL_SetError("Couldn't create audio thread");
+            return 0;
+        }
+    }
+
+    return id + 1;
+}
+
+
+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 {
+        id = open_audio_device(NULL, 0, desired, desired, 0, 1);
+    }
+    if (id > 1) {               /* this should never happen in theory... */
+        SDL_CloseAudioDevice(id);
+        SDL_SetError("Internal error"); /* MUST be Device ID #1! */
+        return (-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 && device->enabled) {
+        if (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) {
+        device->paused = pause_on;
+    }
+}
+
+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)
+{
+    SDL_AudioDevice *device = get_audio_device(devid);
+    if (device) {
+        close_audio_device(device);
+        open_devices[devid - 1] = NULL;
+    }
+}
+
+void
+SDL_CloseAudio(void)
+{
+    SDL_CloseAudioDevice(1);
+}
+
+void
+SDL_AudioQuit(void)
+{
+    SDL_AudioDeviceID i;
+    for (i = 0; i < SDL_arraysize(open_devices); i++) {
+        SDL_CloseAudioDevice(i);
+    }
+
+    /* Free the driver data */
+    current_audio.impl.Deinitialize();
+    SDL_memset(&current_audio, '\0', sizeof(current_audio));
+    SDL_memset(open_devices, '\0', sizeof(open_devices));
+}
+
+#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_AudioFormat format;
+        if (device->convert.needed) {
+            format = device->convert.src_format;
+        } else {
+            format = device->spec.format;
+        }
+        SDL_MixAudioFormat(dst, src, format, len, volume);
+    }
+}
+
+/* vi: set ts=4 sw=4 expandtab: */