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Added SDL 1c0f6952e65e.

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This commit is contained in:
rude
2015-03-15 15:47:43 +01:00
parent 4d04ac181e
commit 6a2aff6450
1109 changed files with 160136 additions and 33995 deletions
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libs/SDL2/src/audio/SDL_audio.c
+261 -30
View File
@@ -51,6 +51,7 @@ extern AudioBootStrap QSAAUDIO_bootstrap;
extern AudioBootStrap SUNAUDIO_bootstrap;
extern AudioBootStrap ARTS_bootstrap;
extern AudioBootStrap ESD_bootstrap;
extern AudioBootStrap NACLAUD_bootstrap;
extern AudioBootStrap NAS_bootstrap;
extern AudioBootStrap XAUDIO2_bootstrap;
extern AudioBootStrap DSOUND_bootstrap;
@@ -68,6 +69,8 @@ extern AudioBootStrap FUSIONSOUND_bootstrap;
extern AudioBootStrap ANDROIDAUD_bootstrap;
extern AudioBootStrap PSPAUD_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
extern AudioBootStrap EmscriptenAudio_bootstrap;
/* Available audio drivers */
static const AudioBootStrap *const bootstrap[] = {
@@ -98,6 +101,9 @@ static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_ESD
&ESD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_NACL
&NACLAUD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_NAS
&NAS_bootstrap,
#endif
@@ -133,6 +139,9 @@ static const AudioBootStrap *const bootstrap[] = {
#endif
#if SDL_AUDIO_DRIVER_PSP
&PSPAUD_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_EMSCRIPTEN
&EmscriptenAudio_bootstrap,
#endif
NULL
};
@@ -171,6 +180,12 @@ SDL_AudioPlayDevice_Default(_THIS)
{ /* no-op. */
}
static int
SDL_AudioGetPendingBytes_Default(_THIS)
{
return 0;
}
static Uint8 *
SDL_AudioGetDeviceBuf_Default(_THIS)
{
@@ -198,22 +213,34 @@ SDL_AudioOpenDevice_Default(_THIS, const char *devname, int iscapture)
return -1;
}
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 (device->thread && (SDL_ThreadID() == device->threadid)) {
return;
if (!is_in_audio_device_thread(device)) {
SDL_LockMutex(device->mixer_lock);
}
SDL_LockMutex(device->mixer_lock);
}
static void
SDL_AudioUnlockDevice_Default(SDL_AudioDevice * device)
{
if (device->thread && (SDL_ThreadID() == device->threadid)) {
return;
if (!is_in_audio_device_thread(device)) {
SDL_UnlockMutex(device->mixer_lock);
}
SDL_UnlockMutex(device->mixer_lock);
}
@@ -234,6 +261,7 @@ finalize_audio_entry_points(void)
FILL_STUB(ThreadInit);
FILL_STUB(WaitDevice);
FILL_STUB(PlayDevice);
FILL_STUB(GetPendingBytes);
FILL_STUB(GetDeviceBuf);
FILL_STUB(WaitDone);
FILL_STUB(CloseDevice);
@@ -243,6 +271,7 @@ finalize_audio_entry_points(void)
#undef FILL_STUB
}
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
/* Streaming functions (for when the input and output buffer sizes are different) */
/* Write [length] bytes from buf into the streamer */
static void
@@ -302,8 +331,184 @@ SDL_StreamDeinit(SDL_AudioStreamer * stream)
{
SDL_free(stream->buffer);
}
#endif
#if defined(ANDROID)
/* buffer queueing support... */
/* this expects that you managed thread safety elsewhere. */
static void
free_audio_queue(SDL_AudioBufferQueue *buffer)
{
while (buffer) {
SDL_AudioBufferQueue *next = buffer->next;
SDL_free(buffer);
buffer = next;
}
}
static void SDLCALL
SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int _len)
{
/* this function always holds the mixer lock before being called. */
Uint32 len = (Uint32) _len;
SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
SDL_AudioBufferQueue *buffer;
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(_len >= 0); /* this shouldn't ever happen, right?! */
while ((len > 0) && ((buffer = device->buffer_queue_head) != NULL)) {
const Uint32 avail = buffer->datalen - buffer->startpos;
const Uint32 cpy = SDL_min(len, avail);
SDL_assert(device->queued_bytes >= avail);
SDL_memcpy(stream, buffer->data + buffer->startpos, cpy);
buffer->startpos += cpy;
stream += cpy;
device->queued_bytes -= cpy;
len -= cpy;
if (buffer->startpos == buffer->datalen) { /* packet is done, put it in the pool. */
device->buffer_queue_head = buffer->next;
SDL_assert((buffer->next != NULL) || (buffer == device->buffer_queue_tail));
buffer->next = device->buffer_queue_pool;
device->buffer_queue_pool = buffer;
}
}
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
if (len > 0) { /* fill any remaining space in the stream with silence. */
SDL_assert(device->buffer_queue_head == NULL);
SDL_memset(stream, device->spec.silence, len);
}
if (device->buffer_queue_head == NULL) {
device->buffer_queue_tail = NULL; /* in case we drained the queue entirely. */
}
}
int
SDL_QueueAudio(SDL_AudioDeviceID devid, const void *_data, Uint32 len)
{
SDL_AudioDevice *device = get_audio_device(devid);
const Uint8 *data = (const Uint8 *) _data;
SDL_AudioBufferQueue *orighead;
SDL_AudioBufferQueue *origtail;
Uint32 origlen;
Uint32 datalen;
if (!device) {
return -1; /* get_audio_device() will have set the error state */
}
if (device->spec.callback != SDL_BufferQueueDrainCallback) {
return SDL_SetError("Audio device has a callback, queueing not allowed");
}
current_audio.impl.LockDevice(device);
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;
current_audio.impl.UnlockDevice(device);
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;
}
current_audio.impl.UnlockDevice(device);
return 0;
}
Uint32
SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
{
Uint32 retval = 0;
SDL_AudioDevice *device = get_audio_device(devid);
/* Nothing to do unless we're set up for queueing. */
if (device && (device->spec.callback == SDL_BufferQueueDrainCallback)) {
current_audio.impl.LockDevice(device);
retval = device->queued_bytes + current_audio.impl.GetPendingBytes(device);
current_audio.impl.UnlockDevice(device);
}
return retval;
}
void
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
{
SDL_AudioDevice *device = get_audio_device(devid);
SDL_AudioBufferQueue *buffer = NULL;
if (!device) {
return; /* nothing to do. */
}
/* Blank out the device and release the mutex. Free it afterwards. */
current_audio.impl.LockDevice(device);
buffer = device->buffer_queue_head;
device->buffer_queue_tail = NULL;
device->buffer_queue_head = NULL;
device->queued_bytes = 0;
current_audio.impl.UnlockDevice(device);
free_audio_queue(buffer);
}
#if defined(__ANDROID__)
#include <android/log.h>
#endif
@@ -317,9 +522,12 @@ SDL_RunAudio(void *devicep)
void *udata;
void (SDLCALL * fill) (void *userdata, Uint8 * stream, int len);
Uint32 delay;
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
/* For streaming when the buffer sizes don't match up */
Uint8 *istream;
int istream_len = 0;
#endif
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
@@ -367,6 +575,7 @@ SDL_RunAudio(void *devicep)
delay = ((device->spec.samples * 1000) / device->spec.freq);
/* Determine if the streamer is necessary here */
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
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,
@@ -455,7 +664,9 @@ SDL_RunAudio(void *devicep)
}
}
} else {
} else
#endif
{
/* Otherwise, do not use the streamer. This is the old code. */
const int silence = (int) device->spec.silence;
@@ -510,10 +721,12 @@ SDL_RunAudio(void *devicep)
current_audio.impl.WaitDone(device);
/* If necessary, deinit the streamer */
#if 0 /* !!! FIXME: rewrite/remove this streamer code. */
if (device->use_streamer == 1)
SDL_StreamDeinit(&device->streamer);
#endif
return (0);
return 0;
}
@@ -546,16 +759,16 @@ SDL_ParseAudioFormat(const char *string)
int
SDL_GetNumAudioDrivers(void)
{
return (SDL_arraysize(bootstrap) - 1);
return SDL_arraysize(bootstrap) - 1;
}
const char *
SDL_GetAudioDriver(int index)
{
if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
return (bootstrap[index]->name);
return bootstrap[index]->name;
}
return (NULL);
return NULL;
}
int
@@ -603,12 +816,12 @@ SDL_AudioInit(const char *driver_name)
}
SDL_memset(&current_audio, 0, sizeof(current_audio));
return (-1); /* No driver was available, so fail. */
return -1; /* No driver was available, so fail. */
}
finalize_audio_entry_points();
return (0);
return 0;
}
/*
@@ -771,6 +984,10 @@ close_audio_device(SDL_AudioDevice * device)
current_audio.impl.CloseDevice(device);
device->opened = 0;
}
free_audio_queue(device->buffer_queue_head);
free_audio_queue(device->buffer_queue_pool);
SDL_FreeAudioMem(device);
}
@@ -785,11 +1002,6 @@ 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)) {
@@ -842,7 +1054,6 @@ prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
return 1;
}
static SDL_AudioDeviceID
open_audio_device(const char *devname, int iscapture,
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
@@ -921,7 +1132,7 @@ open_audio_device(const char *devname, int iscapture,
SDL_OutOfMemory();
return 0;
}
SDL_memset(device, '\0', sizeof(SDL_AudioDevice));
SDL_zerop(device);
device->spec = *obtained;
device->enabled = 1;
device->paused = 1;
@@ -939,8 +1150,9 @@ open_audio_device(const char *devname, int iscapture,
/* force a device detection if we haven't done one yet. */
if ( ((iscapture) && (current_audio.inputDevices == NULL)) ||
((!iscapture) && (current_audio.outputDevices == NULL)) )
((!iscapture) && (current_audio.outputDevices == NULL)) ) {
SDL_GetNumAudioDevices(iscapture);
}
if (current_audio.impl.OpenDevice(device, devname, iscapture) < 0) {
close_audio_device(device);
@@ -1014,6 +1226,25 @@ open_audio_device(const char *devname, int iscapture,
}
}
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->spec.callback = SDL_BufferQueueDrainCallback;
device->spec.userdata = device;
}
/* 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) {
@@ -1063,25 +1294,25 @@ SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
/* 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);
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);
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);
id = open_audio_device(NULL, 0, desired, NULL, 0, 1);
}
SDL_assert((id == 0) || (id == 1));
return ((id == 0) ? -1 : 0);
return (id == 0) ? -1 : 0;
}
SDL_AudioDeviceID
@@ -1105,7 +1336,7 @@ SDL_GetAudioDeviceStatus(SDL_AudioDeviceID devid)
status = SDL_AUDIO_PLAYING;
}
}
return (status);
return status;
}
@@ -1241,16 +1472,16 @@ SDL_FirstAudioFormat(SDL_AudioFormat format)
}
}
format_idx_sub = 0;
return (SDL_NextAudioFormat());
return SDL_NextAudioFormat();
}
SDL_AudioFormat
SDL_NextAudioFormat(void)
{
if ((format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS)) {
return (0);
return 0;
}
return (format_list[format_idx][format_idx_sub++]);
return format_list[format_idx][format_idx_sub++];
}
void
libs/SDL2/src/audio/SDL_audiotypecvt.c
+100 -100
View File
@@ -2300,7 +2300,7 @@ SDL_Upsample_U8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 1;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -2332,7 +2332,7 @@ SDL_Downsample_U8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@@ -2364,7 +2364,7 @@ SDL_Upsample_U8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 2;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -2401,7 +2401,7 @@ SDL_Downsample_U8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@@ -2438,7 +2438,7 @@ SDL_Upsample_U8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 4;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -2485,7 +2485,7 @@ SDL_Downsample_U8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@@ -2532,7 +2532,7 @@ SDL_Upsample_U8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 6;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -2589,7 +2589,7 @@ SDL_Downsample_U8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@@ -2646,7 +2646,7 @@ SDL_Upsample_U8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 8;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -2713,7 +2713,7 @@ SDL_Downsample_U8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@@ -2780,7 +2780,7 @@ SDL_Upsample_S8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 1;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -2812,7 +2812,7 @@ SDL_Downsample_S8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@@ -2844,7 +2844,7 @@ SDL_Upsample_S8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 2;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -2881,7 +2881,7 @@ SDL_Downsample_S8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@@ -2918,7 +2918,7 @@ SDL_Upsample_S8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 4;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -2965,7 +2965,7 @@ SDL_Downsample_S8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@@ -3012,7 +3012,7 @@ SDL_Upsample_S8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 6;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -3069,7 +3069,7 @@ SDL_Downsample_S8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@@ -3126,7 +3126,7 @@ SDL_Upsample_S8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 8;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -3193,7 +3193,7 @@ SDL_Downsample_S8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@@ -3260,7 +3260,7 @@ SDL_Upsample_U16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -3292,7 +3292,7 @@ SDL_Downsample_U16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -3324,7 +3324,7 @@ SDL_Upsample_U16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -3361,7 +3361,7 @@ SDL_Downsample_U16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -3398,7 +3398,7 @@ SDL_Upsample_U16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -3445,7 +3445,7 @@ SDL_Downsample_U16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -3492,7 +3492,7 @@ SDL_Upsample_U16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -3549,7 +3549,7 @@ SDL_Downsample_U16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -3606,7 +3606,7 @@ SDL_Upsample_U16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -3673,7 +3673,7 @@ SDL_Downsample_U16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -3740,7 +3740,7 @@ SDL_Upsample_S16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -3772,7 +3772,7 @@ SDL_Downsample_S16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -3804,7 +3804,7 @@ SDL_Upsample_S16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -3841,7 +3841,7 @@ SDL_Downsample_S16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -3878,7 +3878,7 @@ SDL_Upsample_S16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -3925,7 +3925,7 @@ SDL_Downsample_S16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -3972,7 +3972,7 @@ SDL_Upsample_S16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -4029,7 +4029,7 @@ SDL_Downsample_S16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -4086,7 +4086,7 @@ SDL_Upsample_S16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -4153,7 +4153,7 @@ SDL_Downsample_S16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -4220,7 +4220,7 @@ SDL_Upsample_U16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -4252,7 +4252,7 @@ SDL_Downsample_U16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -4284,7 +4284,7 @@ SDL_Upsample_U16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -4321,7 +4321,7 @@ SDL_Downsample_U16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -4358,7 +4358,7 @@ SDL_Upsample_U16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -4405,7 +4405,7 @@ SDL_Downsample_U16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -4452,7 +4452,7 @@ SDL_Upsample_U16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -4509,7 +4509,7 @@ SDL_Downsample_U16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -4566,7 +4566,7 @@ SDL_Upsample_U16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -4633,7 +4633,7 @@ SDL_Downsample_U16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@@ -4700,7 +4700,7 @@ SDL_Upsample_S16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -4732,7 +4732,7 @@ SDL_Downsample_S16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -4764,7 +4764,7 @@ SDL_Upsample_S16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -4801,7 +4801,7 @@ SDL_Downsample_S16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -4838,7 +4838,7 @@ SDL_Upsample_S16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -4885,7 +4885,7 @@ SDL_Downsample_S16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -4932,7 +4932,7 @@ SDL_Upsample_S16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -4989,7 +4989,7 @@ SDL_Downsample_S16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -5046,7 +5046,7 @@ SDL_Upsample_S16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -5113,7 +5113,7 @@ SDL_Downsample_S16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@@ -5180,7 +5180,7 @@ SDL_Upsample_S32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -5212,7 +5212,7 @@ SDL_Downsample_S32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5244,7 +5244,7 @@ SDL_Upsample_S32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -5281,7 +5281,7 @@ SDL_Downsample_S32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5318,7 +5318,7 @@ SDL_Upsample_S32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -5365,7 +5365,7 @@ SDL_Downsample_S32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5412,7 +5412,7 @@ SDL_Upsample_S32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -5469,7 +5469,7 @@ SDL_Downsample_S32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5526,7 +5526,7 @@ SDL_Upsample_S32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -5593,7 +5593,7 @@ SDL_Downsample_S32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5660,7 +5660,7 @@ SDL_Upsample_S32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -5692,7 +5692,7 @@ SDL_Downsample_S32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5724,7 +5724,7 @@ SDL_Upsample_S32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -5761,7 +5761,7 @@ SDL_Downsample_S32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5798,7 +5798,7 @@ SDL_Upsample_S32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -5845,7 +5845,7 @@ SDL_Downsample_S32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -5892,7 +5892,7 @@ SDL_Upsample_S32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -5949,7 +5949,7 @@ SDL_Downsample_S32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -6006,7 +6006,7 @@ SDL_Upsample_S32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -6073,7 +6073,7 @@ SDL_Downsample_S32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@@ -6140,7 +6140,7 @@ SDL_Upsample_F32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 1;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -6172,7 +6172,7 @@ SDL_Downsample_F32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6204,7 +6204,7 @@ SDL_Upsample_F32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 2;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -6241,7 +6241,7 @@ SDL_Downsample_F32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6278,7 +6278,7 @@ SDL_Upsample_F32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 4;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -6325,7 +6325,7 @@ SDL_Downsample_F32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6372,7 +6372,7 @@ SDL_Upsample_F32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 6;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -6429,7 +6429,7 @@ SDL_Downsample_F32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6486,7 +6486,7 @@ SDL_Upsample_F32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 8;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -6553,7 +6553,7 @@ SDL_Downsample_F32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6620,7 +6620,7 @@ SDL_Upsample_F32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 1;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1;
@@ -6652,7 +6652,7 @@ SDL_Downsample_F32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6684,7 +6684,7 @@ SDL_Upsample_F32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 2;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2;
@@ -6721,7 +6721,7 @@ SDL_Downsample_F32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6758,7 +6758,7 @@ SDL_Upsample_F32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 4;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4;
@@ -6805,7 +6805,7 @@ SDL_Downsample_F32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6852,7 +6852,7 @@ SDL_Upsample_F32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 6;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6;
@@ -6909,7 +6909,7 @@ SDL_Downsample_F32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@@ -6966,7 +6966,7 @@ SDL_Upsample_F32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 8;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8;
@@ -7033,7 +7033,7 @@ SDL_Downsample_F32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
libs/SDL2/src/audio/SDL_sysaudio.h
+27
View File
@@ -33,6 +33,26 @@ typedef struct SDL_AudioDevice SDL_AudioDevice;
/* Used by audio targets during DetectDevices() */
typedef void (*SDL_AddAudioDevice)(const char *name);
/* This is the size of a packet when using SDL_QueueAudio(). We allocate
these as necessary and pool them, under the assumption that we'll
eventually end up with a handful that keep recycling, meeting whatever
the app needs. We keep packing data tightly as more arrives to avoid
wasting space, and if we get a giant block of data, we'll split them
into multiple packets behind the scenes. My expectation is that most
apps will have 2-3 of these in the pool. 8k should cover most needs, but
if this is crippling for some embedded system, we can #ifdef this.
The system preallocates enough packets for 2 callbacks' worth of data. */
#define SDL_AUDIOBUFFERQUEUE_PACKETLEN (8 * 1024)
/* Used by apps that queue audio instead of using the callback. */
typedef struct SDL_AudioBufferQueue
{
Uint8 data[SDL_AUDIOBUFFERQUEUE_PACKETLEN]; /* packet data. */
Uint32 datalen; /* bytes currently in use in this packet. */
Uint32 startpos; /* bytes currently consumed in this packet. */
struct SDL_AudioBufferQueue *next; /* next item in linked list. */
} SDL_AudioBufferQueue;
typedef struct SDL_AudioDriverImpl
{
void (*DetectDevices) (int iscapture, SDL_AddAudioDevice addfn);
@@ -40,6 +60,7 @@ typedef struct SDL_AudioDriverImpl
void (*ThreadInit) (_THIS); /* Called by audio thread at start */
void (*WaitDevice) (_THIS);
void (*PlayDevice) (_THIS);
int (*GetPendingBytes) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
void (*WaitDone) (_THIS);
void (*CloseDevice) (_THIS);
@@ -119,6 +140,12 @@ struct SDL_AudioDevice
SDL_Thread *thread;
SDL_threadID threadid;
/* Queued buffers (if app not using callback). */
SDL_AudioBufferQueue *buffer_queue_head; /* device fed from here. */
SDL_AudioBufferQueue *buffer_queue_tail; /* queue fills to here. */
SDL_AudioBufferQueue *buffer_queue_pool; /* these are unused packets. */
Uint32 queued_bytes; /* number of bytes of audio data in the queue. */
/* * * */
/* Data private to this driver */
struct SDL_PrivateAudioData *hidden;
libs/SDL2/src/audio/SDL_wave.c
+8 -7
View File
@@ -121,7 +121,8 @@ MS_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
struct MS_ADPCM_decodestate *state[2];
Uint8 *freeable, *encoded, *decoded;
Sint32 encoded_len, samplesleft;
Sint8 nybble, stereo;
Sint8 nybble;
Uint8 stereo;
Sint16 *coeff[2];
Sint32 new_sample;
@@ -278,7 +279,8 @@ IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state, Uint8 nybble)
} else if (state->index < 0) {
state->index = 0;
}
step = step_table[state->index];
/* explicit cast to avoid gcc warning about using 'char' as array index */
step = step_table[(int)state->index];
delta = step >> 3;
if (nybble & 0x04)
delta += step;
@@ -343,8 +345,8 @@ IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
/* Check to make sure we have enough variables in the state array */
channels = IMA_ADPCM_state.wavefmt.channels;
if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
SDL_SetError("IMA ADPCM decoder can only handle %d channels",
SDL_arraysize(IMA_ADPCM_state.state));
SDL_SetError("IMA ADPCM decoder can only handle %u channels",
(unsigned int)SDL_arraysize(IMA_ADPCM_state.state));
return (-1);
}
state = IMA_ADPCM_state.state;
@@ -458,7 +460,7 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
}
/* 2 Uint32's for chunk header+len, plus the lenread */
headerDiff += lenread + 2 * sizeof(Uint32);
} while ((chunk.magic == FACT) || (chunk.magic == LIST));
} while ((chunk.magic == FACT) || (chunk.magic == LIST) || (chunk.magic == BEXT) || (chunk.magic == JUNK));
/* Decode the audio data format */
format = (WaveFMT *) chunk.data;
@@ -493,8 +495,7 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
IMA_ADPCM_encoded = 1;
break;
case MP3_CODE:
SDL_SetError("MPEG Layer 3 data not supported",
SDL_SwapLE16(format->encoding));
SDL_SetError("MPEG Layer 3 data not supported");
was_error = 1;
goto done;
default:
libs/SDL2/src/audio/SDL_wave.h
+2
View File
@@ -29,6 +29,8 @@
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define BEXT 0x74786562 /* "bext" */
#define JUNK 0x4B4E554A /* "JUNK" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
#define PCM_CODE 0x0001
libs/SDL2/src/audio/android/SDL_androidaudio.c
+45 -1
View File
@@ -32,7 +32,7 @@
#include <android/log.h>
static void * audioDevice;
static SDL_AudioDevice* audioDevice = NULL;
static int
AndroidAUD_OpenDevice(_THIS, const char *devname, int iscapture)
@@ -49,6 +49,11 @@ AndroidAUD_OpenDevice(_THIS, const char *devname, int iscapture)
}
audioDevice = this;
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) { /* no "UNKNOWN" constant */
@@ -110,6 +115,10 @@ AndroidAUD_CloseDevice(_THIS)
Android_JNI_CloseAudioDevice();
if (audioDevice == this) {
if (audioDevice->hidden != NULL) {
SDL_free(this->hidden);
this->hidden = NULL;
}
audioDevice = NULL;
}
}
@@ -135,6 +144,41 @@ AudioBootStrap ANDROIDAUD_bootstrap = {
"android", "SDL Android audio driver", AndroidAUD_Init, 0
};
/* Pause (block) all non already paused audio devices by taking their mixer lock */
void AndroidAUD_PauseDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (audioDevice->paused) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
}
else {
SDL_LockMutex(audioDevice->mixer_lock);
audioDevice->paused = SDL_TRUE;
private->resume = SDL_TRUE;
}
}
}
/* Resume (unblock) all non already paused audio devices by releasing their mixer lock */
void AndroidAUD_ResumeDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (private->resume) {
audioDevice->paused = SDL_FALSE;
private->resume = SDL_FALSE;
SDL_UnlockMutex(audioDevice->mixer_lock);
}
}
}
#endif /* SDL_AUDIO_DRIVER_ANDROID */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/android/SDL_androidaudio.h
+2
View File
@@ -30,6 +30,8 @@
struct SDL_PrivateAudioData
{
/* Resume device if it was paused automatically */
int resume;
};
static void AndroidAUD_CloseDevice(_THIS);
libs/SDL2/src/audio/coreaudio/SDL_coreaudio.c
+9 -6
View File
@@ -19,6 +19,9 @@
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_COREAUDIO
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
@@ -319,7 +322,6 @@ COREAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->audioUnitOpened) {
OSStatus result = noErr;
AURenderCallbackStruct callback;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
@@ -331,14 +333,13 @@ COREAUDIO_CloseDevice(_THIS)
kAudioUnitScope_Input);
/* stop processing the audio unit */
result = AudioOutputUnitStop(this->hidden->audioUnit);
AudioOutputUnitStop(this->hidden->audioUnit);
/* Remove the input callback */
SDL_memset(&callback, 0, sizeof(AURenderCallbackStruct));
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback,
sizeof(callback));
AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
#if MACOSX_COREAUDIO
CloseComponent(this->hidden->audioUnit);
@@ -556,4 +557,6 @@ AudioBootStrap COREAUDIO_bootstrap = {
"coreaudio", "CoreAudio", COREAUDIO_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_COREAUDIO */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/directsound/SDL_directsound.h
+1 -1
View File
@@ -23,7 +23,7 @@
#ifndef _SDL_directsound_h
#define _SDL_directsound_h
#include "directx.h"
#include "../../core/windows/SDL_directx.h"
#include "../SDL_sysaudio.h"
libs/SDL2/src/audio/directsound/directx.h
-102
View File
@@ -1,102 +0,0 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef _directx_h
#define _directx_h
/* Include all of the DirectX 8.0 headers and adds any necessary tweaks */
#include "../../core/windows/SDL_windows.h"
#include <mmsystem.h>
#ifndef WIN32
#define WIN32
#endif
#undef WINNT
/* Far pointers don't exist in 32-bit code */
#ifndef FAR
#define FAR
#endif
/* Error codes not yet included in Win32 API header files */
#ifndef MAKE_HRESULT
#define MAKE_HRESULT(sev,fac,code) \
((HRESULT)(((unsigned long)(sev)<<31) | ((unsigned long)(fac)<<16) | ((unsigned long)(code))))
#endif
#ifndef S_OK
#define S_OK (HRESULT)0x00000000L
#endif
#ifndef SUCCEEDED
#define SUCCEEDED(x) ((HRESULT)(x) >= 0)
#endif
#ifndef FAILED
#define FAILED(x) ((HRESULT)(x)<0)
#endif
#ifndef E_FAIL
#define E_FAIL (HRESULT)0x80000008L
#endif
#ifndef E_NOINTERFACE
#define E_NOINTERFACE (HRESULT)0x80004002L
#endif
#ifndef E_OUTOFMEMORY
#define E_OUTOFMEMORY (HRESULT)0x8007000EL
#endif
#ifndef E_INVALIDARG
#define E_INVALIDARG (HRESULT)0x80070057L
#endif
#ifndef E_NOTIMPL
#define E_NOTIMPL (HRESULT)0x80004001L
#endif
#ifndef REGDB_E_CLASSNOTREG
#define REGDB_E_CLASSNOTREG (HRESULT)0x80040154L
#endif
/* Severity codes */
#ifndef SEVERITY_ERROR
#define SEVERITY_ERROR 1
#endif
/* Error facility codes */
#ifndef FACILITY_WIN32
#define FACILITY_WIN32 7
#endif
#ifndef FIELD_OFFSET
#define FIELD_OFFSET(type, field) ((LONG)&(((type *)0)->field))
#endif
/* DirectX headers (if it isn't included, I haven't tested it yet)
*/
/* We need these defines to mark what version of DirectX API we use */
#define DIRECTDRAW_VERSION 0x0700
#define DIRECTSOUND_VERSION 0x0800
#define DIRECTINPUT_VERSION 0x0500
#include <ddraw.h>
#include <dsound.h>
#include <dinput.h>
#endif /* _directx_h */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/emscripten/SDL_emscriptenaudio.c
+275
View File
@@ -0,0 +1,275 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_EMSCRIPTEN
#include "SDL_audio.h"
#include "SDL_log.h"
#include "../SDL_audio_c.h"
#include "SDL_emscriptenaudio.h"
#include <emscripten/emscripten.h>
static int
copyData(_THIS)
{
int byte_len;
if (this->hidden->write_off + this->convert.len_cvt > this->hidden->mixlen) {
if (this->hidden->write_off > this->hidden->read_off) {
SDL_memmove(this->hidden->mixbuf,
this->hidden->mixbuf + this->hidden->read_off,
this->hidden->mixlen - this->hidden->read_off);
this->hidden->write_off = this->hidden->write_off - this->hidden->read_off;
} else {
this->hidden->write_off = 0;
}
this->hidden->read_off = 0;
}
SDL_memcpy(this->hidden->mixbuf + this->hidden->write_off,
this->convert.buf,
this->convert.len_cvt);
this->hidden->write_off += this->convert.len_cvt;
byte_len = this->hidden->write_off - this->hidden->read_off;
return byte_len;
}
static void
HandleAudioProcess(_THIS)
{
Uint8 *buf = NULL;
int byte_len = 0;
int bytes = SDL_AUDIO_BITSIZE(this->spec.format) / 8;
int bytes_in = SDL_AUDIO_BITSIZE(this->convert.src_format) / 8;
/* Only do soemthing if audio is enabled */
if (!this->enabled)
return;
if (this->paused)
return;
if (this->convert.needed) {
if (this->hidden->conv_in_len != 0) {
this->convert.len = this->hidden->conv_in_len * bytes_in * this->spec.channels;
}
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
buf = this->convert.buf;
byte_len = this->convert.len_cvt;
/* size mismatch*/
if (byte_len != this->spec.size) {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size > byte_len ? this->spec.size * 2 : byte_len * 2;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
}
/* copy existing data */
byte_len = copyData(this);
/* read more data*/
while (byte_len < this->spec.size) {
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
byte_len = copyData(this);
}
byte_len = this->spec.size;
buf = this->hidden->mixbuf + this->hidden->read_off;
this->hidden->read_off += byte_len;
}
} else {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
}
(*this->spec.callback) (this->spec.userdata,
this->hidden->mixbuf,
this->hidden->mixlen);
buf = this->hidden->mixbuf;
byte_len = this->hidden->mixlen;
}
if (buf) {
EM_ASM_ARGS({
var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
if (channelData.length != $1) {
throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
channelData[j] = getValue($0 + (j*numChannels + c)*4, 'float');
}
}
}, buf, byte_len / bytes / this->spec.channels);
}
}
static void
Emscripten_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
/* Clean up the audio buffer */
SDL_free(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
Emscripten_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_bool valid_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int i;
float f;
int result;
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_F32: /* web audio only supports floats */
this->spec.format = test_format;
valid_format = SDL_TRUE;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
/* Didn't find a compatible format :( */
return SDL_SetError("No compatible audio format!");
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* based on parts of library_sdl.js */
/* create context (TODO: this puts stuff in the global namespace...)*/
result = EM_ASM_INT_V({
if(typeof(SDL2) === 'undefined')
SDL2 = {};
if(typeof(SDL2.audio) === 'undefined')
SDL2.audio = {};
if (!SDL2.audioContext) {
if (typeof(AudioContext) !== 'undefined') {
SDL2.audioContext = new AudioContext();
} else if (typeof(webkitAudioContext) !== 'undefined') {
SDL2.audioContext = new webkitAudioContext();
} else {
return -1;
}
}
return 0;
});
if (result < 0) {
return SDL_SetError("Web Audio API is not available!");
}
/* limit to native freq */
int sampleRate = EM_ASM_INT_V({
return SDL2.audioContext['sampleRate'];
});
if(this->spec.freq != sampleRate) {
for (i = this->spec.samples; i > 0; i--) {
f = (float)i / (float)sampleRate * (float)this->spec.freq;
if (SDL_floor(f) == f) {
this->hidden->conv_in_len = SDL_floor(f);
break;
}
}
this->spec.freq = sampleRate;
}
SDL_CalculateAudioSpec(&this->spec);
/* setup a ScriptProcessorNode */
EM_ASM_ARGS({
SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
SDL2.audio.currentOutputBuffer = e['outputBuffer'];
Runtime.dynCall('vi', $2, [$3]);
};
SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
return 0;
}
static int
Emscripten_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = Emscripten_OpenDevice;
impl->CloseDevice = Emscripten_CloseDevice;
/* only one output */
impl->OnlyHasDefaultOutputDevice = 1;
/* no threads here */
impl->SkipMixerLock = 1;
impl->ProvidesOwnCallbackThread = 1;
/* check availability */
int available = EM_ASM_INT_V({
if (typeof(AudioContext) !== 'undefined') {
return 1;
} else if (typeof(webkitAudioContext) !== 'undefined') {
return 1;
}
return 0;
});
return available;
}
AudioBootStrap EmscriptenAudio_bootstrap = {
"emscripten", "SDL emscripten audio driver", Emscripten_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/emscripten/SDL_emscriptenaudio.h
+42
View File
@@ -0,0 +1,42 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#ifndef _SDL_emscriptenaudio_h
#define _SDL_emscriptenaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 conv_in_len;
Uint32 write_off, read_off;
};
#endif /* _SDL_emscriptenaudio_h */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/nacl/SDL_naclaudio.c
+154
View File
@@ -0,0 +1,154 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_NACL
#include "SDL_naclaudio.h"
#include "SDL_audio.h"
#include "SDL_mutex.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "ppapi/c/pp_errors.h"
#include "ppapi/c/pp_instance.h"
#include "ppapi_simple/ps.h"
#include "ppapi_simple/ps_interface.h"
#include "ppapi_simple/ps_event.h"
/* The tag name used by NACL audio */
#define NACLAUD_DRIVER_NAME "nacl"
#define SAMPLE_FRAME_COUNT 4096
/* Audio driver functions */
static int NACLAUD_OpenDevice(_THIS, const char *devname, int iscapture);
static void NACLAUD_CloseDevice(_THIS);
static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data);
/* FIXME: Make use of latency if needed */
static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data) {
SDL_AudioDevice* _this = (SDL_AudioDevice*) data;
SDL_LockMutex(private->mutex);
if (_this->enabled && !_this->paused) {
if (_this->convert.needed) {
SDL_LockMutex(_this->mixer_lock);
(*_this->spec.callback) (_this->spec.userdata,
(Uint8 *) _this->convert.buf,
_this->convert.len);
SDL_UnlockMutex(_this->mixer_lock);
SDL_ConvertAudio(&_this->convert);
SDL_memcpy(samples, _this->convert.buf, _this->convert.len_cvt);
} else {
SDL_LockMutex(_this->mixer_lock);
(*_this->spec.callback) (_this->spec.userdata, (Uint8 *) samples, buffer_size);
SDL_UnlockMutex(_this->mixer_lock);
}
} else {
SDL_memset(samples, 0, buffer_size);
}
return;
}
static void NACLAUD_CloseDevice(SDL_AudioDevice *device) {
const PPB_Core *core = PSInterfaceCore();
const PPB_Audio *ppb_audio = PSInterfaceAudio();
SDL_PrivateAudioData *hidden = (SDL_PrivateAudioData *) device->hidden;
ppb_audio->StopPlayback(hidden->audio);
SDL_DestroyMutex(hidden->mutex);
core->ReleaseResource(hidden->audio);
}
static int
NACLAUD_OpenDevice(_THIS, const char *devname, int iscapture) {
PP_Instance instance = PSGetInstanceId();
const PPB_Audio *ppb_audio = PSInterfaceAudio();
const PPB_AudioConfig *ppb_audiocfg = PSInterfaceAudioConfig();
private = (SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *private));
if (private == NULL) {
SDL_OutOfMemory();
return 0;
}
private->mutex = SDL_CreateMutex();
_this->spec.freq = 44100;
_this->spec.format = AUDIO_S16LSB;
_this->spec.channels = 2;
_this->spec.samples = ppb_audiocfg->RecommendSampleFrameCount(
instance,
PP_AUDIOSAMPLERATE_44100,
SAMPLE_FRAME_COUNT);
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&_this->spec);
private->audio = ppb_audio->Create(
instance,
ppb_audiocfg->CreateStereo16Bit(instance, PP_AUDIOSAMPLERATE_44100, _this->spec.samples),
nacl_audio_callback,
_this);
/* Start audio playback while we are still on the main thread. */
ppb_audio->StartPlayback(private->audio);
return 1;
}
static int
NACLAUD_Init(SDL_AudioDriverImpl * impl)
{
if (PSGetInstanceId() == 0) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = NACLAUD_OpenDevice;
impl->CloseDevice = NACLAUD_CloseDevice;
impl->HasCaptureSupport = 0;
impl->OnlyHasDefaultOutputDevice = 1;
impl->OnlyHasDefaultInputDevice = 1;
impl->ProvidesOwnCallbackThread = 1;
/*
* impl->WaitDevice = NACLAUD_WaitDevice;
* impl->GetDeviceBuf = NACLAUD_GetDeviceBuf;
* impl->PlayDevice = NACLAUD_PlayDevice;
* impl->Deinitialize = NACLAUD_Deinitialize;
*/
return 1;
}
AudioBootStrap NACLAUD_bootstrap = {
NACLAUD_DRIVER_NAME, "SDL NaCl Audio Driver",
NACLAUD_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_NACL */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/nacl/SDL_naclaudio.h
+41
View File
@@ -0,0 +1,41 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#ifndef _SDL_naclaudio_h
#define _SDL_naclaudio_h
#include "SDL_audio.h"
#include "../SDL_sysaudio.h"
#include "SDL_mutex.h"
#include "ppapi/c/ppb_audio.h"
#define _THIS SDL_AudioDevice *_this
#define private _this->hidden
typedef struct SDL_PrivateAudioData {
SDL_mutex* mutex;
PP_Resource audio;
} SDL_PrivateAudioData;
#endif /* _SDL_naclaudio_h */
libs/SDL2/src/audio/psp/SDL_pspaudio.c
+5 -1
View File
@@ -18,6 +18,9 @@
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_PSP
#include <stdio.h>
#include <string.h>
@@ -191,5 +194,6 @@ AudioBootStrap PSPAUD_bootstrap = {
/* SDL_AUDI */
#endif /* SDL_AUDIO_DRIVER_PSP */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/psp/SDL_pspaudio.h
+1 -1
View File
@@ -24,7 +24,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the video functions */
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2
libs/SDL2/src/audio/sdlgenaudiocvt.pl
+2 -1
View File
@@ -383,6 +383,7 @@ sub buildArbitraryResampleFunc {
my $eps_adjust = ($upsample) ? 'dstsize' : 'srcsize';
my $incr = '';
my $incr2 = '';
my $block_align = $channels * $fsize/8;
# !!! FIXME: DEBUG_CONVERT should report frequencies.
@@ -395,7 +396,7 @@ ${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - $fudge;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/${block_align})) * cvt->rate_incr) * ${block_align};
register int eps = 0;
EOF
libs/SDL2/src/audio/sndio/SDL_sndioaudio.c
+20 -15
View File
@@ -209,7 +209,7 @@ SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
this->hidden->mixlen = this->spec.size;
/* !!! FIXME: SIO_DEVANY can be a specific device... */
if ((this->hidden->dev = SNDIO_sio_open(NULL, SIO_PLAY, 0)) == NULL) {
if ((this->hidden->dev = SNDIO_sio_open(SIO_DEVANY, SIO_PLAY, 0)) == NULL) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_open() failed");
}
@@ -229,7 +229,17 @@ SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
par.sig = SDL_AUDIO_ISSIGNED(test_format) ? 1 : 0;
par.bits = SDL_AUDIO_BITSIZE(test_format);
if (SNDIO_sio_setpar(this->hidden->dev, &par) == 1) {
if (SNDIO_sio_setpar(this->hidden->dev, &par) == 0) {
continue;
}
if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_getpar() failed");
}
if (par.bps != SIO_BPS(par.bits)) {
continue;
}
if ((par.bits == 8 * par.bps) || (par.msb)) {
status = 0;
break;
}
@@ -242,26 +252,21 @@ SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
return SDL_SetError("sndio: Couldn't find any hardware audio formats");
}
if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_getpar() failed");
}
if ((par.bits == 32) && (par.sig) && (par.le))
if ((par.bps == 4) && (par.sig) && (par.le))
this->spec.format = AUDIO_S32LSB;
else if ((par.bits == 32) && (par.sig) && (!par.le))
else if ((par.bps == 4) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S32MSB;
else if ((par.bits == 16) && (par.sig) && (par.le))
else if ((par.bps == 2) && (par.sig) && (par.le))
this->spec.format = AUDIO_S16LSB;
else if ((par.bits == 16) && (par.sig) && (!par.le))
else if ((par.bps == 2) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S16MSB;
else if ((par.bits == 16) && (!par.sig) && (par.le))
else if ((par.bps == 2) && (!par.sig) && (par.le))
this->spec.format = AUDIO_U16LSB;
else if ((par.bits == 16) && (!par.sig) && (!par.le))
else if ((par.bps == 2) && (!par.sig) && (!par.le))
this->spec.format = AUDIO_U16MSB;
else if ((par.bits == 8) && (par.sig))
else if ((par.bps == 1) && (par.sig))
this->spec.format = AUDIO_S8;
else if ((par.bits == 8) && (!par.sig))
else if ((par.bps == 1) && (!par.sig))
this->spec.format = AUDIO_U8;
else {
SNDIO_CloseDevice(this);
libs/SDL2/src/audio/xaudio2/SDL_xaudio2.c
+3 -3
View File
@@ -58,7 +58,7 @@
/* The configure script already did any necessary checking */
# define SDL_XAUDIO2_HAS_SDK 1
#elif defined(__WINRT__)
/* WinRT always has access to the .the XAudio 2 SDK */
/* WinRT always has access to the XAudio 2 SDK */
# define SDL_XAUDIO2_HAS_SDK
#else
/* XAudio2 exists as of the March 2008 DirectX SDK
@@ -241,14 +241,14 @@ XAUDIO2_WaitDone(_THIS)
SDL_assert(!this->enabled); /* flag that stops playing. */
IXAudio2SourceVoice_Discontinuity(source);
#if SDL_XAUDIO2_WIN8
IXAudio2SourceVoice_GetState(source, &state, 0);
IXAudio2SourceVoice_GetState(source, &state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
#else
IXAudio2SourceVoice_GetState(source, &state);
#endif
while (state.BuffersQueued > 0) {
SDL_SemWait(this->hidden->semaphore);
#if SDL_XAUDIO2_WIN8
IXAudio2SourceVoice_GetState(source, &state, 0);
IXAudio2SourceVoice_GetState(source, &state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
#else
IXAudio2SourceVoice_GetState(source, &state);
#endif