updae source to 2.0.9 source

Edward Rudd

2019-04-09 9cd2e9ec8fc0127393dfce9c0359d500c8c238be

 source/src/audio/SDL_audio.c

@@ -1,6 +1,6 @@
/*
  Simple DirectMedia Layer
  Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
  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
@@ -33,36 +33,6 @@
static SDL_AudioDriver current_audio;
static SDL_AudioDevice *open_devices[16];

/*
 * 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 PULSEAUDIO_bootstrap;
extern AudioBootStrap ALSA_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
extern AudioBootStrap BSD_AUDIO_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 XAUDIO2_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 ANDROIDAUDIO_bootstrap;
extern AudioBootStrap PSPAUDIO_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;

/* Available audio drivers */
static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_PULSEAUDIO
@@ -74,8 +44,8 @@
#if SDL_AUDIO_DRIVER_SNDIO
    &SNDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_BSD
    &BSD_AUDIO_bootstrap,
#if SDL_AUDIO_DRIVER_NETBSD
    &NETBSDAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_OSS
    &DSP_bootstrap,
@@ -98,8 +68,8 @@
#if SDL_AUDIO_DRIVER_NAS
    &NAS_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_XAUDIO2
    &XAUDIO2_bootstrap,
#if SDL_AUDIO_DRIVER_WASAPI
    &WASAPI_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DSOUND
    &DSOUND_bootstrap,
@@ -116,12 +86,6 @@
#if SDL_AUDIO_DRIVER_COREAUDIO
    &COREAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DISK
    &DISKAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DUMMY
    &DUMMYAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_FUSIONSOUND
    &FUSIONSOUND_bootstrap,
#endif
@@ -134,8 +98,100 @@
#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)
@@ -166,6 +222,16 @@

static void
SDL_AudioThreadInit_Default(_THIS)
{                               /* no-op. */
}

static void
SDL_AudioThreadDeinit_Default(_THIS)
{                               /* no-op. */
}

static void
SDL_AudioBeginLoopIteration_Default(_THIS)
{                               /* no-op. */
}

@@ -288,6 +354,8 @@
    FILL_STUB(DetectDevices);
    FILL_STUB(OpenDevice);
    FILL_STUB(ThreadInit);
    FILL_STUB(ThreadDeinit);
    FILL_STUB(BeginLoopIteration);
    FILL_STUB(WaitDevice);
    FILL_STUB(PlayDevice);
    FILL_STUB(GetPendingBytes);
@@ -310,21 +378,57 @@
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_AudioDeviceItem *item;
    const SDL_AudioDeviceItem *i;
    int dupenum = 0;

    SDL_assert(handle != NULL);  /* we reserve NULL, audio backends can't use it. */
    SDL_assert(name != NULL);

    item = (SDL_AudioDeviceItem *) SDL_malloc(sizeof (SDL_AudioDeviceItem));
    if (!item) {
        return SDL_OutOfMemory();
    }

    item->original_name = SDL_strdup(name);
    if (!item->original_name) {
        SDL_free(item);
        return SDL_OutOfMemory();
    }

    item->dupenum = 0;
    item->name = item->original_name;
    item->handle = handle;
    SDL_strlcpy(item->name, name, size - sizeof (SDL_AudioDeviceItem));

    SDL_LockMutex(current_audio.detectionLock);

    for (i = *devices; i != NULL; i = i->next) {
        if (SDL_strcmp(name, i->original_name) == 0) {
            dupenum = i->dupenum + 1;
            break;  /* stop at the highest-numbered dupe. */
        }
    }

    if (dupenum) {
        const size_t len = SDL_strlen(name) + 16;
        char *replacement = (char *) SDL_malloc(len);
        if (!replacement) {
            SDL_UnlockMutex(current_audio.detectionLock);
            SDL_free(item->original_name);
            SDL_free(item);
            SDL_OutOfMemory();
            return -1;
        }

        SDL_snprintf(replacement, len, "%s (%d)", name, dupenum + 1);
        item->dupenum = dupenum;
        item->name = replacement;
    }

    item->next = *devices;
    *devices = item;
    retval = (*devCount)++;
    retval = (*devCount)++;   /* !!! FIXME: this should be an atomic increment */

    SDL_UnlockMutex(current_audio.detectionLock);

    return retval;
@@ -352,6 +456,11 @@
        if (item->handle != NULL) {
            current_audio.impl.FreeDeviceHandle(item->handle);
        }
        /* these two pointers are the same if not a duplicate devname */
        if (item->name != item->original_name) {
            SDL_free(item->name);
        }
        SDL_free(item->original_name);
        SDL_free(item);
    }
    *devices = NULL;
@@ -383,7 +492,11 @@
    SDL_assert(get_audio_device(device->id) == device);

    if (!SDL_AtomicGet(&device->enabled)) {
        return;
        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
@@ -448,136 +561,23 @@

/* buffer queueing support... */

/* this expects that you managed thread safety elsewhere. */
static void
free_audio_queue(SDL_AudioBufferQueue *packet)
{
    while (packet) {
        SDL_AudioBufferQueue *next = packet->next;
        SDL_free(packet);
        packet = next;
    }
}

/* NOTE: This assumes you'll hold the mixer lock before calling! */
static int
queue_audio_to_device(SDL_AudioDevice *device, const Uint8 *data, Uint32 len)
{
    SDL_AudioBufferQueue *orighead;
    SDL_AudioBufferQueue *origtail;
    Uint32 origlen;
    Uint32 datalen;

    orighead = device->buffer_queue_head;
    origtail = device->buffer_queue_tail;
    origlen = origtail ? origtail->datalen : 0;

    while (len > 0) {
        SDL_AudioBufferQueue *packet = device->buffer_queue_tail;
        SDL_assert(!packet || (packet->datalen <= SDL_AUDIOBUFFERQUEUE_PACKETLEN));
        if (!packet || (packet->datalen >= SDL_AUDIOBUFFERQUEUE_PACKETLEN)) {
            /* tail packet missing or completely full; we need a new packet. */
            packet = device->buffer_queue_pool;
            if (packet != NULL) {
                /* we have one available in the pool. */
                device->buffer_queue_pool = packet->next;
            } else {
                /* Have to allocate a new one! */
                packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
                if (packet == NULL) {
                    /* uhoh, reset so we've queued nothing new, free what we can. */
                    if (!origtail) {
                        packet = device->buffer_queue_head;  /* whole queue. */
                    } else {
                        packet = origtail->next;  /* what we added to existing queue. */
                        origtail->next = NULL;
                        origtail->datalen = origlen;
                    }
                    device->buffer_queue_head = orighead;
                    device->buffer_queue_tail = origtail;
                    device->buffer_queue_pool = NULL;

                    free_audio_queue(packet);  /* give back what we can. */

                    return SDL_OutOfMemory();
                }
            }
            packet->datalen = 0;
            packet->startpos = 0;
            packet->next = NULL;

            SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
            if (device->buffer_queue_tail == NULL) {
                device->buffer_queue_head = packet;
            } else {
                device->buffer_queue_tail->next = packet;
            }
            device->buffer_queue_tail = packet;
        }

        datalen = SDL_min(len, SDL_AUDIOBUFFERQUEUE_PACKETLEN - packet->datalen);
        SDL_memcpy(packet->data + packet->datalen, data, datalen);
        data += datalen;
        len -= datalen;
        packet->datalen += datalen;
        device->queued_bytes += datalen;
    }

    return 0;
}

/* NOTE: This assumes you'll hold the mixer lock before calling! */
static Uint32
dequeue_audio_from_device(SDL_AudioDevice *device, Uint8 *stream, Uint32 len)
{
    SDL_AudioBufferQueue *packet;
    Uint8 *ptr = stream;

    while ((len > 0) && ((packet = device->buffer_queue_head) != NULL)) {
        const Uint32 avail = packet->datalen - packet->startpos;
        const Uint32 cpy = SDL_min(len, avail);
        SDL_assert(device->queued_bytes >= avail);

        SDL_memcpy(ptr, packet->data + packet->startpos, cpy);
        packet->startpos += cpy;
        ptr += cpy;
        device->queued_bytes -= cpy;
        len -= cpy;

        if (packet->startpos == packet->datalen) {  /* packet is done, put it in the pool. */
            device->buffer_queue_head = packet->next;
            SDL_assert((packet->next != NULL) || (packet == device->buffer_queue_tail));
            packet->next = device->buffer_queue_pool;
            device->buffer_queue_pool = packet;
        }
    }

    SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));

    if (device->buffer_queue_head == NULL) {
        device->buffer_queue_tail = NULL;  /* in case we drained the queue entirely. */
    }

    return (Uint32) (ptr - stream);
}

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;
    Uint32 written;
    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?! */

    written = dequeue_audio_from_device(device, stream, (Uint32) len);
    stream += written;
    len -= (int) written;
    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(device->buffer_queue_head == NULL);
        SDL_assert(SDL_CountDataQueue(device->buffer_queue) == 0);
        SDL_memset(stream, device->spec.silence, len);
    }
}
@@ -595,7 +595,7 @@
    /* 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. */
    queue_audio_to_device(device, stream, (Uint32) len);
    SDL_WriteToDataQueue(device->buffer_queue, stream, len);
}

int
@@ -608,13 +608,13 @@
        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->spec.callback != SDL_BufferQueueDrainCallback) {
    } 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 = queue_audio_to_device(device, data, len);
        rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
        current_audio.impl.UnlockDevice(device);
    }

@@ -630,12 +630,12 @@
    if ( (len == 0) ||  /* nothing to do? */
         (!device) ||  /* called with bogus device id */
         (!device->iscapture) ||  /* playback devices can't dequeue */
         (device->spec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
         (device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
        return 0;  /* just report zero bytes dequeued. */
    }

    current_audio.impl.LockDevice(device);
    rc = dequeue_audio_from_device(device, data, len);
    rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
    current_audio.impl.UnlockDevice(device);
    return rc;
}
@@ -651,13 +651,13 @@
    }

    /* Nothing to do unless we're set up for queueing. */
    if (device->spec.callback == SDL_BufferQueueDrainCallback) {
    if (device->callbackspec.callback == SDL_BufferQueueDrainCallback) {
        current_audio.impl.LockDevice(device);
        retval = device->queued_bytes + current_audio.impl.GetPendingBytes(device);
        retval = ((Uint32) SDL_CountDataQueue(device->buffer_queue)) + current_audio.impl.GetPendingBytes(device);
        current_audio.impl.UnlockDevice(device);
    } else if (device->spec.callback == SDL_BufferQueueFillCallback) {
    } else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
        current_audio.impl.LockDevice(device);
        retval = device->queued_bytes;
        retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
        current_audio.impl.UnlockDevice(device);
    }

@@ -668,7 +668,6 @@
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
{
    SDL_AudioDevice *device = get_audio_device(devid);
    SDL_AudioBufferQueue *packet;

    if (!device) {
        return;  /* nothing to do. */
@@ -677,35 +676,10 @@
    /* Blank out the device and release the mutex. Free it afterwards. */
    current_audio.impl.LockDevice(device);

    /* merge the available pool and the current queue into one list. */
    packet = device->buffer_queue_head;
    if (packet) {
        device->buffer_queue_tail->next = device->buffer_queue_pool;
    } else {
        packet = device->buffer_queue_pool;
    }

    /* Remove the queued packets from the device. */
    device->buffer_queue_tail = NULL;
    device->buffer_queue_head = NULL;
    device->queued_bytes = 0;
    device->buffer_queue_pool = packet;

    /* Keep up to two packets in the pool to reduce future malloc pressure. */
    if (packet) {
        if (!packet->next) {
            packet = NULL;  /* one packet (the only one) for the pool. */
        } else {
            SDL_AudioBufferQueue *next = packet->next->next;
            packet->next->next = NULL;  /* two packets for the pool. */
            packet = next;  /* rest will be freed. */
        }
    }
    SDL_ClearDataQueue(device->buffer_queue, SDL_AUDIOBUFFERQUEUE_PACKETLEN * 2);

    current_audio.impl.UnlockDevice(device);

    /* free any extra packets we didn't keep in the pool. */
    free_audio_queue(packet);
}


@@ -714,17 +688,15 @@
SDL_RunAudio(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 stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
    Uint8 *stream;
    void *udata = device->spec.userdata;
    void (SDLCALL *callback) (void *, Uint8 *, int) = device->spec.callback;
    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_HIGH);
    SDL_SetThreadPriority(SDL_THREAD_PRIORITY_TIME_CRITICAL);

    /* Perform any thread setup */
    device->threadid = SDL_ThreadID();
@@ -732,51 +704,64 @@

    /* 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->convert.needed) {
            stream = device->convert.buf;
        } else if (SDL_AtomicGet(&device->enabled)) {
            stream = current_audio.impl.GetDeviceBuf(device);
        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
               fake_stream, so the app's callback will fire with
               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. */
            stream = NULL;
               now to know if the device failed.
               Streaming playback uses work_buffer, too. */
            data = NULL;
        }

        if (stream == NULL) {
            stream = device->fake_stream;
        if (data == NULL) {
            data = device->work_buffer;
        }

        /* !!! FIXME: this should be LockDevice. */
        if ( SDL_AtomicGet(&device->enabled) ) {
            SDL_LockMutex(device->mixer_lock);
            if (SDL_AtomicGet(&device->paused)) {
                SDL_memset(stream, silence, stream_len);
            } else {
                (*callback) (udata, stream, stream_len);
            }
            SDL_UnlockMutex(device->mixer_lock);
        }

        /* Convert the audio if necessary */
        if (device->convert.needed && SDL_AtomicGet(&device->enabled)) {
            SDL_ConvertAudio(&device->convert);
            stream = current_audio.impl.GetDeviceBuf(device);
            if (stream == NULL) {
                stream = device->fake_stream;
            } else {
                SDL_memcpy(stream, device->convert.buf,
                           device->convert.len_cvt);
            }
        }

        /* Ready current buffer for play and change current buffer */
        if (stream == device->fake_stream) {
            SDL_Delay(delay);
        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);
        }
@@ -787,9 +772,12 @@
    /* 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)
@@ -797,10 +785,10 @@
    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 stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
    Uint8 *stream;
    void *udata = device->spec.userdata;
    void (SDLCALL *callback) (void *, Uint8 *, int) = device->spec.callback;
    const int data_len = device->spec.size;
    Uint8 *data;
    void *udata = device->callbackspec.userdata;
    SDL_AudioCallback callback = device->callbackspec.callback;

    SDL_assert(device->iscapture);

@@ -816,34 +804,43 @@
        int still_need;
        Uint8 *ptr;

        if (!SDL_AtomicGet(&device->enabled) || SDL_AtomicGet(&device->paused)) {
        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 = stream_len;
        if (device->convert.needed) {
            ptr = stream = device->convert.buf;
        } else {
            /* just use the "fake" stream to hold data read from the device. */
            ptr = stream = device->fake_stream;
        }
        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. */

        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 (!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;
                }
            }
        }

@@ -852,21 +849,39 @@
            SDL_memset(ptr, silence, still_need);
        }

        if (device->convert.needed) {
            SDL_ConvertAudio(&device->convert);
        }
        if (device->stream) {
            /* if this fails...oh well. */
            SDL_AudioStreamPut(device->stream, data, data_len);

        /* !!! FIXME: this should be LockDevice. */
        SDL_LockMutex(device->mixer_lock);
        if (SDL_AtomicGet(&device->paused)) {
            current_audio.impl.FlushCapture(device);  /* one snuck in! */
        } else {
            (*callback)(udata, stream, stream_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);
        }
        SDL_UnlockMutex(device->mixer_lock);
    }

    current_audio.impl.PrepareToClose(device);

    current_audio.impl.FlushCapture(device);

    current_audio.impl.ThreadDeinit(device);

    return 0;
}
@@ -968,6 +983,10 @@
    /* Make sure we have a list of devices available at startup. */
    current_audio.impl.DetectDevices();

#ifdef HAVE_LIBSAMPLERATE_H
    LoadLibSampleRate();
#endif

    return 0;
}

@@ -999,6 +1018,11 @@
            } else {
                *devices = next;
            }
            /* these two pointers are the same if not a duplicate devname */
            if (item->name != item->original_name) {
                SDL_free(item->name);
            }
            SDL_free(item->original_name);
            SDL_free(item);
        }
        item = next;
@@ -1025,7 +1049,6 @@

    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;
@@ -1082,6 +1105,24 @@
        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));
@@ -1090,24 +1131,11 @@
        }
    }

    SDL_AtomicSet(&device->shutdown, 1);
    SDL_AtomicSet(&device->enabled, 0);
    if (device->thread != NULL) {
        SDL_WaitThread(device->thread, NULL);
    }
    if (device->mixer_lock != NULL) {
        SDL_DestroyMutex(device->mixer_lock);
    }
    SDL_free(device->fake_stream);
    if (device->convert.needed) {
        SDL_free(device->convert.buf);
    }
    if (device->hidden != NULL) {
        current_audio.impl.CloseDevice(device);
    }

    free_audio_queue(device->buffer_queue_head);
    free_audio_queue(device->buffer_queue_pool);
    SDL_FreeDataQueue(device->buffer_queue);

    SDL_free(device);
}
@@ -1147,8 +1175,9 @@
        }
    case 1:                    /* Mono */
    case 2:                    /* Stereo */
    case 4:                    /* surround */
    case 6:                    /* surround with center and lfe */
    case 4:                    /* Quadrophonic */
    case 6:                    /* 5.1 surround */
    case 8:                    /* 7.1 surround */
        break;
    default:
        SDL_SetError("Unsupported number of audio channels.");
@@ -1180,11 +1209,11 @@
                  const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
                  int allowed_changes, int min_id)
{
    const SDL_bool is_internal_thread = (desired->callback != NULL);
    const SDL_bool is_internal_thread = (desired->callback == NULL);
    SDL_AudioDeviceID id = 0;
    SDL_AudioSpec _obtained;
    SDL_AudioDevice *device;
    SDL_bool build_cvt;
    SDL_bool build_stream;
    void *handle = NULL;
    int i = 0;

@@ -1319,84 +1348,84 @@
    SDL_assert(device->hidden != NULL);

    /* See if we need to do any conversion */
    build_cvt = SDL_FALSE;
    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_cvt = SDL_TRUE;
            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_cvt = SDL_TRUE;
            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_cvt = SDL_TRUE;
            build_stream = SDL_TRUE;
        }
    }
    if (device->spec.samples != obtained->samples) {
        if (allowed_changes & SDL_AUDIO_ALLOW_SAMPLES_CHANGE) {
            obtained->samples = device->spec.samples;
        } else {
            build_stream = SDL_TRUE;
        }
    }

    /* If the audio driver changes the buffer size, accept it.
       This needs to be done after the format is modified above,
       otherwise it might not have the correct buffer size.
     */
    if (device->spec.samples != obtained->samples) {
        obtained->samples = device->spec.samples;
        SDL_CalculateAudioSpec(obtained);
    }
    SDL_CalculateAudioSpec(obtained);  /* recalc after possible changes. */

    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) {
    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->convert.needed) {
            device->convert.len = (int) (((double) device->spec.size) /
                                         device->convert.len_ratio);

            device->convert.buf =
                (Uint8 *) SDL_malloc(device->convert.len *
                                            device->convert.len_mult);
            if (device->convert.buf == NULL) {
                close_audio_device(device);
                SDL_OutOfMemory();
                return 0;
            }
        }
    }

    if (device->spec.callback == NULL) {  /* use buffer queueing? */
        /* pool a few packets to start. Enough for two callbacks. */
        const int packetlen = SDL_AUDIOBUFFERQUEUE_PACKETLEN;
        const int wantbytes = ((device->convert.needed) ? device->convert.len : device->spec.size) * 2;
        const int wantpackets = (wantbytes / packetlen) + ((wantbytes % packetlen) ? packetlen : 0);
        for (i = 0; i < wantpackets; i++) {
            SDL_AudioBufferQueue *packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
            if (packet) { /* don't care if this fails, we'll deal later. */
                packet->datalen = 0;
                packet->startpos = 0;
                packet->next = device->buffer_queue_pool;
                device->buffer_queue_pool = packet;
            }
        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->spec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback;
        device->spec.userdata = device;
        device->callbackspec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback;
        device->callbackspec.userdata = device;
    }

    /* add it to our list of open devices. */
    open_devices[id] = 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) {
@@ -1406,21 +1435,7 @@
        const size_t stacksize = is_internal_thread ? 64 * 1024 : 0;
        char threadname[64];

        /* Allocate a fake audio buffer; only used by our internal threads. */
        Uint32 stream_len = (device->convert.needed) ? device->convert.len_cvt : 0;
        if (device->spec.size > stream_len) {
            stream_len = device->spec.size;
        }
        SDL_assert(stream_len > 0);

        device->fake_stream = (Uint8 *) SDL_malloc(stream_len);
        if (device->fake_stream == NULL) {
            close_audio_device(device);
            SDL_OutOfMemory();
            return 0;
        }

        SDL_snprintf(threadname, sizeof (threadname), "SDLAudioDev%d", (int) device->id);
        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) {
@@ -1456,7 +1471,14 @@
        id = open_audio_device(NULL, 0, desired, obtained,
                               SDL_AUDIO_ALLOW_ANY_CHANGE, 1);
    } else {
        id = open_audio_device(NULL, 0, desired, NULL, 0, 1);
        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));
@@ -1579,6 +1601,12 @@

    SDL_zero(current_audio);
    SDL_zero(open_devices);

#ifdef HAVE_LIBSAMPLERATE_H
    UnloadLibSampleRate();
#endif

    SDL_FreeResampleFilter();
}

#define NUM_FORMATS 10
@@ -1655,13 +1683,7 @@
    /* 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);
        SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
    }
}