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Revert "Updated SDL, Bullet and OpenAL soft libs"

Browse source 
This reverts commit 370161cfb1.
This commit is contained in:
AzaezelX 2019-07-08 09:49:44 -05:00
parent 63be684474
commit bc77ff0833
1102 changed files with 62741 additions and 204988 deletions
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111
Engine/lib/sdl/src/audio/SDL_audio.c
View file

@ -378,57 +378,21 @@ static int
add_audio_device(const char *name, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
{
int retval = -1;
SDL_AudioDeviceItem *item;
const SDL_AudioDeviceItem *i;
int dupenum = 0;
const size_t size = sizeof (SDL_AudioDeviceItem) + SDL_strlen(name) + 1;
SDL_AudioDeviceItem *item = (SDL_AudioDeviceItem *) SDL_malloc(size);
if (item == NULL) {
return -1;
}
SDL_assert(handle != NULL); /* we reserve NULL, audio backends can't use it. */
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)++; /* !!! FIXME: this should be an atomic increment */
retval = (*devCount)++;
SDL_UnlockMutex(current_audio.detectionLock);
return retval;
@ -456,11 +420,6 @@ free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
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;
@ -492,11 +451,7 @@ void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
SDL_assert(get_audio_device(device->id) == device);
if (!SDL_AtomicGet(&device->enabled)) {
return; /* don't report disconnects more than once. */
}
if (SDL_AtomicGet(&device->shutdown)) {
return; /* don't report disconnect if we're trying to close device. */
return;
}
/* Ends the audio callback and mark the device as STOPPED, but the
@ -696,7 +651,7 @@ SDL_RunAudio(void *devicep)
SDL_assert(!device->iscapture);
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_TIME_CRITICAL);
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
/* Perform any thread setup */
device->threadid = SDL_ThreadID();
@ -877,8 +832,6 @@ SDL_CaptureAudio(void *devicep)
}
}
current_audio.impl.PrepareToClose(device);
current_audio.impl.FlushCapture(device);
current_audio.impl.ThreadDeinit(device);
@ -1018,11 +971,6 @@ clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *re
} 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;
@ -1049,6 +997,7 @@ SDL_GetNumAudioDevices(int iscapture)
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;
@ -1105,14 +1054,16 @@ close_audio_device(SDL_AudioDevice * device)
return;
}
/* make sure the device is paused before we do anything else, so the
audio callback definitely won't fire again. */
current_audio.impl.LockDevice(device);
SDL_AtomicSet(&device->paused, 1);
if (device->id > 0) {
SDL_AudioDevice *opendev = open_devices[device->id - 1];
SDL_assert((opendev == device) || (opendev == NULL));
if (opendev == device) {
open_devices[device->id - 1] = NULL;
}
}
SDL_AtomicSet(&device->shutdown, 1);
SDL_AtomicSet(&device->enabled, 0);
current_audio.impl.UnlockDevice(device);
if (device->thread != NULL) {
SDL_WaitThread(device->thread, NULL);
}
@ -1123,14 +1074,6 @@ close_audio_device(SDL_AudioDevice * device)
SDL_free(device->work_buffer);
SDL_FreeAudioStream(device->stream);
if (device->id > 0) {
SDL_AudioDevice *opendev = open_devices[device->id - 1];
SDL_assert((opendev == device) || (opendev == NULL));
if (opendev == device) {
open_devices[device->id - 1] = NULL;
}
}
if (device->hidden != NULL) {
current_audio.impl.CloseDevice(device);
}
@ -1175,9 +1118,8 @@ prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
}
case 1: /* Mono */
case 2: /* Stereo */
case 4: /* Quadrophonic */
case 6: /* 5.1 surround */
case 8: /* 7.1 surround */
case 4: /* surround */
case 6: /* surround with center and lfe */
break;
default:
SDL_SetError("Unsupported number of audio channels.");
@ -1370,12 +1312,15 @@ open_audio_device(const char *devname, int iscapture,
build_stream = SDL_TRUE;
}
}
/* !!! FIXME in 2.1: add SDL_AUDIO_ALLOW_SAMPLES_CHANGE flag?
As of 2.0.6, we will build a stream to buffer the difference between
what the app wants to feed and the device wants to eat, so everyone
gets their way. In prior releases, SDL would force the callback to
feed at the rate the device requested, adjusted for resampling.
*/
if (device->spec.samples != obtained->samples) {
if (allowed_changes & SDL_AUDIO_ALLOW_SAMPLES_CHANGE) {
obtained->samples = device->spec.samples;
} else {
build_stream = SDL_TRUE;
}
build_stream = SDL_TRUE;
}
SDL_CalculateAudioSpec(obtained); /* recalc after possible changes. */

4
Engine/lib/sdl/src/audio/SDL_audiocvt.c
View file

@ -724,7 +724,7 @@ SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format
SDL_assert(format == AUDIO_F32SYS);
/* we keep no streaming state here, so pad with silence on both ends. */
padding = (float *) SDL_calloc(paddingsamples ? paddingsamples : 1, sizeof (float));
padding = (float *) SDL_calloc(paddingsamples, sizeof (float));
if (!padding) {
SDL_OutOfMemory();
return;
@ -1291,7 +1291,7 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
retval->packetlen = packetlen;
retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels;
retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples ? retval->resampler_padding_samples : 1, sizeof (float));
retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples, sizeof (float));
if (retval->resampler_padding == NULL) {
SDL_FreeAudioStream(retval);

6
Engine/lib/sdl/src/audio/SDL_audiodev_c.h
View file

@ -18,10 +18,6 @@
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef SDL_audiodev_c_h_
#define SDL_audiodev_c_h_
#include "SDL.h"
#include "../SDL_internal.h"
#include "SDL_sysaudio.h"
@ -39,6 +35,4 @@
extern void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int));
#endif /* SDL_audiodev_c_h_ */
/* vi: set ts=4 sw=4 expandtab: */

736
Engine/lib/sdl/src/audio/SDL_audiotypecvt.c
View file

@ -25,10 +25,8 @@
#include "SDL_cpuinfo.h"
#include "SDL_assert.h"
/* !!! FIXME: disabled until we fix https://bugzilla.libsdl.org/show_bug.cgi?id=4186 */
#if 0 /*def __ARM_NEON__*/
#define HAVE_NEON_INTRINSICS 1
#endif
/* !!! FIXME: write NEON code. */
#define HAVE_NEON_INTRINSICS 0
#ifdef __SSE2__
#define HAVE_SSE2_INTRINSICS 1
@ -64,7 +62,7 @@ SDL_AudioFilter SDL_Convert_F32_to_S32 = NULL;
#define DIVBY128 0.0078125f
#define DIVBY32768 0.000030517578125f
#define DIVBY8388607 0.00000011920930376163766f
#define DIVBY2147483648 0.00000000046566128730773926
#if NEED_SCALAR_CONVERTER_FALLBACKS
@ -154,7 +152,7 @@ SDL_Convert_S32_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32");
for (i = cvt->len_cvt / sizeof (Sint32); i; --i, ++src, ++dst) {
*dst = ((float) (*src>>8)) * DIVBY8388607;
*dst = (float) (((double) *src) * DIVBY2147483648);
}
if (cvt->filters[++cvt->filter_index]) {
@ -173,10 +171,10 @@ SDL_Convert_F32_to_S8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
if (sample > 1.0f) {
*dst = 127;
} else if (sample <= -1.0f) {
*dst = -128;
} else if (sample < -1.0f) {
*dst = -127;
} else {
*dst = (Sint8)(sample * 127.0f);
}
@ -199,9 +197,9 @@ SDL_Convert_F32_to_U8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
if (sample > 1.0f) {
*dst = 255;
} else if (sample <= -1.0f) {
} else if (sample < -1.0f) {
*dst = 0;
} else {
*dst = (Uint8)((sample + 1.0f) * 127.0f);
@ -225,10 +223,10 @@ SDL_Convert_F32_to_S16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
if (sample > 1.0f) {
*dst = 32767;
} else if (sample <= -1.0f) {
*dst = -32768;
} else if (sample < -1.0f) {
*dst = -32767;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
@ -251,9 +249,9 @@ SDL_Convert_F32_to_U16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 65535;
} else if (sample <= -1.0f) {
if (sample > 1.0f) {
*dst = 65534;
} else if (sample < -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
@ -277,12 +275,12 @@ SDL_Convert_F32_to_S32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format)
for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
if (sample > 1.0f) {
*dst = 2147483647;
} else if (sample <= -1.0f) {
*dst = (Sint32) -2147483648LL;
} else if (sample < -1.0f) {
*dst = -2147483647;
} else {
*dst = ((Sint32)(sample * 8388607.0f)) << 8;
*dst = (Sint32)((double)sample * 2147483647.0);
}
}
@ -511,6 +509,16 @@ SDL_Convert_U16_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
}
}
#if defined(__GNUC__) && (__GNUC__ < 4)
/* these were added as of gcc-4.0: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=19418 */
static inline __m128 _mm_castsi128_ps(__m128i __A) {
return (__m128) __A;
}
static inline __m128i _mm_castps_si128(__m128 __A) {
return (__m128i) __A;
}
#endif
static void SDLCALL
SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
@ -522,7 +530,7 @@ SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (Sint32); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = ((float) (*src>>8)) * DIVBY8388607;
*dst = (float) (((double) *src) * DIVBY2147483648);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -530,11 +538,15 @@ SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 divby8388607 = _mm_set1_ps(DIVBY8388607);
const __m128d divby2147483648 = _mm_set1_pd(DIVBY2147483648);
const __m128i *mmsrc = (const __m128i *) src;
while (i >= 4) { /* 4 * sint32 */
/* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */
_mm_store_ps(dst, _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_load_si128(mmsrc), 8)), divby8388607));
const __m128i ints = _mm_load_si128(mmsrc);
/* bitshift the whole register over, so _mm_cvtepi32_pd can read the top ints in the bottom of the vector. */
const __m128d doubles1 = _mm_mul_pd(_mm_cvtepi32_pd(_mm_srli_si128(ints, 8)), divby2147483648);
const __m128d doubles2 = _mm_mul_pd(_mm_cvtepi32_pd(ints), divby2147483648);
/* convert to float32, bitshift/or to get these into a vector to store. */
_mm_store_ps(dst, _mm_castsi128_ps(_mm_or_si128(_mm_slli_si128(_mm_castps_si128(_mm_cvtpd_ps(doubles1)), 8), _mm_castps_si128(_mm_cvtpd_ps(doubles2)))));
i -= 4; mmsrc++; dst += 4;
}
src = (const Sint32 *) mmsrc;
@ -542,7 +554,7 @@ SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = ((float) (*src>>8)) * DIVBY8388607;
*dst = (float) (((double) *src) * DIVBY2147483648);
i--; src++; dst++;
}
@ -562,14 +574,7 @@ SDL_Convert_F32_to_S8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 127;
} else if (sample <= -1.0f) {
*dst = -128;
} else {
*dst = (Sint8)(sample * 127.0f);
}
*dst = (Sint8) (*src * 127.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -577,15 +582,13 @@ SDL_Convert_F32_to_S8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 one = _mm_set1_ps(1.0f);
const __m128 negone = _mm_set1_ps(-1.0f);
const __m128 mulby127 = _mm_set1_ps(127.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 16) { /* 16 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+8)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+12)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+4), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+8), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+12), mulby127)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_packs_epi16(_mm_packs_epi32(ints1, ints2), _mm_packs_epi32(ints3, ints4))); /* pack down, store out. */
i -= 16; src += 16; mmdst++;
}
@ -594,14 +597,7 @@ SDL_Convert_F32_to_S8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 127;
} else if (sample <= -1.0f) {
*dst = -128;
} else {
*dst = (Sint8)(sample * 127.0f);
}
*dst = (Sint8) (*src * 127.0f);
i--; src++; dst++;
}
@ -622,14 +618,7 @@ SDL_Convert_F32_to_U8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 255;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint8)((sample + 1.0f) * 127.0f);
}
*dst = (Uint8) ((*src + 1.0f) * 127.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -637,15 +626,14 @@ SDL_Convert_F32_to_U8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 one = _mm_set1_ps(1.0f);
const __m128 negone = _mm_set1_ps(-1.0f);
const __m128 add1 = _mm_set1_ps(1.0f);
const __m128 mulby127 = _mm_set1_ps(127.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 16) { /* 16 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+8)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+12)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src), add1), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src+4), add1), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src+8), add1), mulby127)); /* load 4 floats, convert to sint32 */
const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_load_ps(src+12), add1), mulby127)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_packus_epi16(_mm_packs_epi32(ints1, ints2), _mm_packs_epi32(ints3, ints4))); /* pack down, store out. */
i -= 16; src += 16; mmdst++;
}
@ -654,14 +642,7 @@ SDL_Convert_F32_to_U8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 255;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint8)((sample + 1.0f) * 127.0f);
}
*dst = (Uint8) ((*src + 1.0f) * 127.0f);
i--; src++; dst++;
}
@ -682,14 +663,7 @@ SDL_Convert_F32_to_S16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 32767;
} else if (sample <= -1.0f) {
*dst = -32768;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
*dst = (Sint16) (*src * 32767.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -697,13 +671,11 @@ SDL_Convert_F32_to_S16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 one = _mm_set1_ps(1.0f);
const __m128 negone = _mm_set1_ps(-1.0f);
const __m128 mulby32767 = _mm_set1_ps(32767.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 8) { /* 8 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src), mulby32767)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+4), mulby32767)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_packs_epi32(ints1, ints2)); /* pack to sint16, store out. */
i -= 8; src += 8; mmdst++;
}
@ -712,14 +684,7 @@ SDL_Convert_F32_to_S16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 32767;
} else if (sample <= -1.0f) {
*dst = -32768;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
*dst = (Sint16) (*src * 32767.0f);
i--; src++; dst++;
}
@ -740,14 +705,7 @@ SDL_Convert_F32_to_U16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 65535;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
}
*dst = (Uint16) ((*src + 1.0f) * 32767.0f);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -764,12 +722,10 @@ SDL_Convert_F32_to_U16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
though it looks like dark magic. */
const __m128 mulby32767 = _mm_set1_ps(32767.0f);
const __m128i topbit = _mm_set1_epi16(-32768);
const __m128 one = _mm_set1_ps(1.0f);
const __m128 negone = _mm_set1_ps(-1.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 8) { /* 8 * float32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src+4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src), mulby32767)); /* load 4 floats, convert to sint32 */
const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_load_ps(src+4), mulby32767)); /* load 4 floats, convert to sint32 */
_mm_store_si128(mmdst, _mm_xor_si128(_mm_packs_epi32(ints1, ints2), topbit)); /* pack to sint16, xor top bit, store out. */
i -= 8; src += 8; mmdst++;
}
@ -778,14 +734,7 @@ SDL_Convert_F32_to_U16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 65535;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
}
*dst = (Uint16) ((*src + 1.0f) * 32767.0f);
i--; src++; dst++;
}
@ -806,14 +755,7 @@ SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 2147483647;
} else if (sample <= -1.0f) {
*dst = (Sint32) -2147483648LL;
} else {
*dst = ((Sint32)(sample * 8388607.0f)) << 8;
}
*dst = (Sint32) (((double) *src) * 2147483647.0);
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -821,12 +763,14 @@ SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
/* Aligned! Do SSE blocks as long as we have 16 bytes available. */
const __m128 one = _mm_set1_ps(1.0f);
const __m128 negone = _mm_set1_ps(-1.0f);
const __m128 mulby8388607 = _mm_set1_ps(8388607.0f);
const __m128d mulby2147483647 = _mm_set1_pd(2147483647.0);
__m128i *mmdst = (__m128i *) dst;
while (i >= 4) { /* 4 * float32 */
_mm_store_si128(mmdst, _mm_slli_epi32(_mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby8388607)), 8)); /* load 4 floats, clamp, convert to sint32 */
const __m128 floats = _mm_load_ps(src);
/* bitshift the whole register over, so _mm_cvtps_pd can read the top floats in the bottom of the vector. */
const __m128d doubles1 = _mm_mul_pd(_mm_cvtps_pd(_mm_castsi128_ps(_mm_srli_si128(_mm_castps_si128(floats), 8))), mulby2147483647);
const __m128d doubles2 = _mm_mul_pd(_mm_cvtps_pd(floats), mulby2147483647);
_mm_store_si128(mmdst, _mm_or_si128(_mm_slli_si128(_mm_cvtpd_epi32(doubles1), 8), _mm_cvtpd_epi32(doubles2)));
i -= 4; src += 4; mmdst++;
}
dst = (Sint32 *) mmdst;
@ -834,14 +778,7 @@ SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 2147483647;
} else if (sample <= -1.0f) {
*dst = (Sint32) -2147483648LL;
} else {
*dst = ((Sint32)(sample * 8388607.0f)) << 8;
}
*dst = (Sint32) (((double) *src) * 2147483647.0);
i--; src++; dst++;
}
@ -852,538 +789,6 @@ SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
#endif
#if HAVE_NEON_INTRINSICS
static void SDLCALL
SDL_Convert_S8_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint8 *src = ((const Sint8 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32 (using NEON)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt; i && (((size_t) (dst-15)) & 15); --i, --src, --dst) {
*dst = ((float) *src) * DIVBY128;
}
src -= 15; dst -= 15; /* adjust to read NEON blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const int8_t *mmsrc = (const int8_t *) src;
const float32x4_t divby128 = vdupq_n_f32(DIVBY128);
while (i >= 16) { /* 16 * 8-bit */
const int8x16_t bytes = vld1q_s8(mmsrc); /* get 16 sint8 into a NEON register. */
const int16x8_t int16hi = vmovl_s8(vget_high_s8(bytes)); /* convert top 8 bytes to 8 int16 */
const int16x8_t int16lo = vmovl_s8(vget_low_s8(bytes)); /* convert bottom 8 bytes to 8 int16 */
/* split int16 to two int32, then convert to float, then multiply to normalize, store. */
vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16hi))), divby128));
vst1q_f32(dst+4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16hi))), divby128));
vst1q_f32(dst+8, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16lo))), divby128));
vst1q_f32(dst+12, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16lo))), divby128));
i -= 16; mmsrc -= 16; dst -= 16;
}
src = (const Sint8 *) mmsrc;
}
src += 15; dst += 15; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = ((float) *src) * DIVBY128;
i--; src--; dst--;
}
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_U8_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Uint8 *src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32 (using NEON)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt; i && (((size_t) (dst-15)) & 15); --i, --src, --dst) {
*dst = (((float) *src) * DIVBY128) - 1.0f;
}
src -= 15; dst -= 15; /* adjust to read NEON blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const uint8_t *mmsrc = (const uint8_t *) src;
const float32x4_t divby128 = vdupq_n_f32(DIVBY128);
const float32x4_t one = vdupq_n_f32(1.0f);
while (i >= 16) { /* 16 * 8-bit */
const uint8x16_t bytes = vld1q_u8(mmsrc); /* get 16 uint8 into a NEON register. */
const uint16x8_t uint16hi = vmovl_u8(vget_high_u8(bytes)); /* convert top 8 bytes to 8 uint16 */
const uint16x8_t uint16lo = vmovl_u8(vget_low_u8(bytes)); /* convert bottom 8 bytes to 8 uint16 */
/* split uint16 to two uint32, then convert to float, then multiply to normalize, subtract to adjust for sign, store. */
vst1q_f32(dst, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16hi))), divby128, one));
vst1q_f32(dst+4, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16hi))), divby128, one));
vst1q_f32(dst+8, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16lo))), divby128, one));
vst1q_f32(dst+12, vmlsq_f32(vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16lo))), divby128, one));
i -= 16; mmsrc -= 16; dst -= 16;
}
src = (const Uint8 *) mmsrc;
}
src += 15; dst += 15; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (((float) *src) * DIVBY128) - 1.0f;
i--; src--; dst--;
}
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_S16_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint16 *src = ((const Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32 (using NEON)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt / sizeof (Sint16); i && (((size_t) (dst-7)) & 15); --i, --src, --dst) {
*dst = ((float) *src) * DIVBY32768;
}
src -= 7; dst -= 7; /* adjust to read NEON blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768);
while (i >= 8) { /* 8 * 16-bit */
const int16x8_t ints = vld1q_s16((int16_t const *) src); /* get 8 sint16 into a NEON register. */
/* split int16 to two int32, then convert to float, then multiply to normalize, store. */
vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(ints))), divby32768));
vst1q_f32(dst+4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(ints))), divby32768));
i -= 8; src -= 8; dst -= 8;
}
}
src += 7; dst += 7; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = ((float) *src) * DIVBY32768;
i--; src--; dst--;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_U16_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Uint16 *src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
float *dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1;
int i;
LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32 (using NEON)");
/* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
for (i = cvt->len_cvt / sizeof (Sint16); i && (((size_t) (dst-7)) & 15); --i, --src, --dst) {
*dst = (((float) *src) * DIVBY32768) - 1.0f;
}
src -= 7; dst -= 7; /* adjust to read NEON blocks from the start. */
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768);
const float32x4_t one = vdupq_n_f32(1.0f);
while (i >= 8) { /* 8 * 16-bit */
const uint16x8_t uints = vld1q_u16((uint16_t const *) src); /* get 8 uint16 into a NEON register. */
/* split uint16 to two int32, then convert to float, then multiply to normalize, subtract for sign, store. */
vst1q_f32(dst, vmlsq_f32(one, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uints))), divby32768));
vst1q_f32(dst+4, vmlsq_f32(one, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uints))), divby32768));
i -= 8; src -= 8; dst -= 8;
}
}
src += 7; dst += 7; /* adjust for any scalar finishing. */
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (((float) *src) * DIVBY32768) - 1.0f;
i--; src--; dst--;
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_S32_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const Sint32 *src = (const Sint32 *) cvt->buf;
float *dst = (float *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32 (using NEON)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (Sint32); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
*dst = ((float) (*src>>8)) * DIVBY8388607;
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
SDL_assert(!i || ((((size_t) src) & 15) == 0));
{
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t divby8388607 = vdupq_n_f32(DIVBY8388607);
const int32_t *mmsrc = (const int32_t *) src;
while (i >= 4) { /* 4 * sint32 */
/* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */
vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vshrq_n_s32(vld1q_s32(mmsrc), 8)), divby8388607));
i -= 4; mmsrc += 4; dst += 4;
}
src = (const Sint32 *) mmsrc;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = ((float) (*src>>8)) * DIVBY8388607;
i--; src++; dst++;
}
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_S8_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint8 *dst = (Sint8 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8 (using NEON)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 127;
} else if (sample <= -1.0f) {
*dst = -128;
} else {
*dst = (Sint8)(sample * 127.0f);
}
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t one = vdupq_n_f32(1.0f);
const float32x4_t negone = vdupq_n_f32(-1.0f);
const float32x4_t mulby127 = vdupq_n_f32(127.0f);
int8_t *mmdst = (int8_t *) dst;
while (i >= 16) { /* 16 * float32 */
const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const int32x4_t ints3 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+8)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const int32x4_t ints4 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+12)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
const int8x8_t i8lo = vmovn_s16(vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2))); /* narrow to sint16, combine, narrow to sint8 */
const int8x8_t i8hi = vmovn_s16(vcombine_s16(vmovn_s32(ints3), vmovn_s32(ints4))); /* narrow to sint16, combine, narrow to sint8 */
vst1q_s8(mmdst, vcombine_s8(i8lo, i8hi)); /* combine to int8x16_t, store out */
i -= 16; src += 16; mmdst += 16;
}
dst = (Sint8 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 127;
} else if (sample <= -1.0f) {
*dst = -128;
} else {
*dst = (Sint8)(sample * 127.0f);
}
i--; src++; dst++;
}
cvt->len_cvt /= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S8);
}
}
static void SDLCALL
SDL_Convert_F32_to_U8_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Uint8 *dst = (Uint8 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8 (using NEON)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 255;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint8)((sample + 1.0f) * 127.0f);
}
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t one = vdupq_n_f32(1.0f);
const float32x4_t negone = vdupq_n_f32(-1.0f);
const float32x4_t mulby127 = vdupq_n_f32(127.0f);
uint8_t *mmdst = (uint8_t *) dst;
while (i >= 16) { /* 16 * float32 */
const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */
const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */
const uint32x4_t uints3 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+8)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */
const uint32x4_t uints4 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+12)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */
const uint8x8_t ui8lo = vmovn_u16(vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2))); /* narrow to uint16, combine, narrow to uint8 */
const uint8x8_t ui8hi = vmovn_u16(vcombine_u16(vmovn_u32(uints3), vmovn_u32(uints4))); /* narrow to uint16, combine, narrow to uint8 */
vst1q_u8(mmdst, vcombine_u8(ui8lo, ui8hi)); /* combine to uint8x16_t, store out */
i -= 16; src += 16; mmdst += 16;
}
dst = (Uint8 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 255;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint8)((sample + 1.0f) * 127.0f);
}
i--; src++; dst++;
}
cvt->len_cvt /= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_U8);
}
}
static void SDLCALL
SDL_Convert_F32_to_S16_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint16 *dst = (Sint16 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16 (using NEON)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 32767;
} else if (sample <= -1.0f) {
*dst = -32768;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t one = vdupq_n_f32(1.0f);
const float32x4_t negone = vdupq_n_f32(-1.0f);
const float32x4_t mulby32767 = vdupq_n_f32(32767.0f);
int16_t *mmdst = (int16_t *) dst;
while (i >= 8) { /* 8 * float32 */
const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
vst1q_s16(mmdst, vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2))); /* narrow to sint16, combine, store out. */
i -= 8; src += 8; mmdst += 8;
}
dst = (Sint16 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 32767;
} else if (sample <= -1.0f) {
*dst = -32768;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
i--; src++; dst++;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_U16_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Uint16 *dst = (Uint16 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16 (using NEON)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 65535;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
}
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
/* Make sure src is aligned too. */
if ((((size_t) src) & 15) == 0) {
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t one = vdupq_n_f32(1.0f);
const float32x4_t negone = vdupq_n_f32(-1.0f);
const float32x4_t mulby32767 = vdupq_n_f32(32767.0f);
uint16_t *mmdst = (uint16_t *) dst;
while (i >= 8) { /* 8 * float32 */
const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby32767)); /* load 4 floats, clamp, convert to uint32 */
const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src+4)), one), one), mulby32767)); /* load 4 floats, clamp, convert to uint32 */
vst1q_u16(mmdst, vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2))); /* narrow to uint16, combine, store out. */
i -= 8; src += 8; mmdst += 8;
}
dst = (Uint16 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 65535;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
}
i--; src++; dst++;
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS);
}
}
static void SDLCALL
SDL_Convert_F32_to_S32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
Sint32 *dst = (Sint32 *) cvt->buf;
int i;
LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32 (using NEON)");
/* Get dst aligned to 16 bytes */
for (i = cvt->len_cvt / sizeof (float); i && (((size_t) dst) & 15); --i, ++src, ++dst) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 2147483647;
} else if (sample <= -1.0f) {
*dst = -2147483648;
} else {
*dst = ((Sint32)(sample * 8388607.0f)) << 8;
}
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
SDL_assert(!i || ((((size_t) src) & 15) == 0));
{
/* Aligned! Do NEON blocks as long as we have 16 bytes available. */
const float32x4_t one = vdupq_n_f32(1.0f);
const float32x4_t negone = vdupq_n_f32(-1.0f);
const float32x4_t mulby8388607 = vdupq_n_f32(8388607.0f);
int32_t *mmdst = (int32_t *) dst;
while (i >= 4) { /* 4 * float32 */
vst1q_s32(mmdst, vshlq_n_s32(vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby8388607)), 8));
i -= 4; src += 4; mmdst += 4;
}
dst = (Sint32 *) mmdst;
}
/* Finish off any leftovers with scalar operations. */
while (i) {
const float sample = *src;
if (sample >= 1.0f) {
*dst = 2147483647;
} else if (sample <= -1.0f) {
*dst = -2147483648;
} else {
*dst = ((Sint32)(sample * 8388607.0f)) << 8;
}
i--; src++; dst++;
}
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS);
}
}
#endif
void SDL_ChooseAudioConverters(void)
{
static SDL_bool converters_chosen = SDL_FALSE;
@ -1412,13 +817,6 @@ void SDL_ChooseAudioConverters(void)
}
#endif
#if HAVE_NEON_INTRINSICS
if (SDL_HasNEON()) {
SET_CONVERTER_FUNCS(NEON);
return;
}
#endif
#if NEED_SCALAR_CONVERTER_FALLBACKS
SET_CONVERTER_FUNCS(Scalar);
#endif

4
Engine/lib/sdl/src/audio/SDL_sysaudio.h
View file

@ -98,10 +98,8 @@ typedef struct SDL_AudioDriverImpl
typedef struct SDL_AudioDeviceItem
{
void *handle;
char *name;
char *original_name;
int dupenum;
struct SDL_AudioDeviceItem *next;
char name[SDL_VARIABLE_LENGTH_ARRAY];
} SDL_AudioDeviceItem;

150
Engine/lib/sdl/src/audio/alsa/SDL_alsa_audio.c
View file

@ -22,10 +22,6 @@
#if SDL_AUDIO_DRIVER_ALSA
#ifndef SDL_ALSA_NON_BLOCKING
#define SDL_ALSA_NON_BLOCKING 0
#endif
/* Allow access to a raw mixing buffer */
#include <sys/types.h>
@ -94,7 +90,6 @@ static int (*ALSA_snd_pcm_reset)(snd_pcm_t *);
static int (*ALSA_snd_device_name_hint) (int, const char *, void ***);
static char* (*ALSA_snd_device_name_get_hint) (const void *, const char *);
static int (*ALSA_snd_device_name_free_hint) (void **);
static snd_pcm_sframes_t (*ALSA_snd_pcm_avail)(snd_pcm_t *);
#ifdef SND_CHMAP_API_VERSION
static snd_pcm_chmap_t* (*ALSA_snd_pcm_get_chmap) (snd_pcm_t *);
static int (*ALSA_snd_pcm_chmap_print) (const snd_pcm_chmap_t *map, size_t maxlen, char *buf);
@ -163,7 +158,6 @@ load_alsa_syms(void)
SDL_ALSA_SYM(snd_device_name_hint);
SDL_ALSA_SYM(snd_device_name_get_hint);
SDL_ALSA_SYM(snd_device_name_free_hint);
SDL_ALSA_SYM(snd_pcm_avail);
#ifdef SND_CHMAP_API_VERSION
SDL_ALSA_SYM(snd_pcm_get_chmap);
SDL_ALSA_SYM(snd_pcm_chmap_print);
@ -249,24 +243,7 @@ get_audio_device(void *handle, const int channels)
static void
ALSA_WaitDevice(_THIS)
{
#if SDL_ALSA_NON_BLOCKING
const snd_pcm_sframes_t needed = (snd_pcm_sframes_t) this->spec.samples;
while (SDL_AtomicGet(&this->enabled)) {
const snd_pcm_sframes_t rc = ALSA_snd_pcm_avail(this->hidden->pcm_handle);
if ((rc < 0) && (rc != -EAGAIN)) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA snd_pcm_avail failed (unrecoverable): %s\n",
ALSA_snd_strerror(rc));
SDL_OpenedAudioDeviceDisconnected(this);
return;
} else if (rc < needed) {
const Uint32 delay = ((needed - (SDL_max(rc, 0))) * 1000) / this->spec.freq;
SDL_Delay(SDL_max(delay, 10));
} else {
break; /* ready to go! */
}
}
#endif
/* We're in blocking mode, so there's nothing to do here */
}
@ -445,7 +422,7 @@ static void
ALSA_CloseDevice(_THIS)
{
if (this->hidden->pcm_handle) {
/* Wait for the submitted audio to drain
/* Wait for the submitted audio to drain
ALSA_snd_pcm_drop() can hang, so don't use that.
*/
Uint32 delay = ((this->spec.samples * 1000) / this->spec.freq) * 2;
@ -458,32 +435,10 @@ ALSA_CloseDevice(_THIS)
}
static int
ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
{
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t bufsize;
snd_pcm_uframes_t persize;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
/* Prioritize matching the period size to the requested buffer size */
persize = this->spec.samples;
status = ALSA_snd_pcm_hw_params_set_period_size_near(
this->hidden->pcm_handle, hwparams, &persize, NULL);
if ( status < 0 ) {
return(-1);
}
/* Next try to restrict the parameters to having only two periods */
bufsize = this->spec.samples * 2;
status = ALSA_snd_pcm_hw_params_set_buffer_size_near(
this->hidden->pcm_handle, hwparams, &bufsize);
if ( status < 0 ) {
return(-1);
}
/* "set" the hardware with the desired parameters */
status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
@ -491,12 +446,24 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
return(-1);
}
this->spec.samples = persize;
/* Get samples for the actual buffer size */
status = ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
if ( status < 0 ) {
return(-1);
}
if ( !override && bufsize != this->spec.samples * 2 ) {
return(-1);
}
/* !!! FIXME: Is this safe to do? */
this->spec.samples = bufsize / 2;
/* This is useful for debugging */
if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {
snd_pcm_uframes_t persize = 0;
unsigned int periods = 0;
ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL);
ALSA_snd_pcm_hw_params_get_periods(hwparams, &periods, NULL);
fprintf(stderr,
@ -507,6 +474,78 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
return(0);
}
static int
ALSA_set_period_size(_THIS, snd_pcm_hw_params_t *params, int override)
{
const char *env;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t frames;
unsigned int periods;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
if ( !override ) {
env = SDL_getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");
if ( env ) {
override = SDL_atoi(env);
if ( override == 0 ) {
return(-1);
}
}
}
frames = this->spec.samples;
status = ALSA_snd_pcm_hw_params_set_period_size_near(
this->hidden->pcm_handle, hwparams, &frames, NULL);
if ( status < 0 ) {
return(-1);
}
periods = 2;
status = ALSA_snd_pcm_hw_params_set_periods_near(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
}
return ALSA_finalize_hardware(this, hwparams, override);
}
static int
ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params, int override)
{
const char *env;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t frames;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
if ( !override ) {
env = SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");
if ( env ) {
override = SDL_atoi(env);
if ( override == 0 ) {
return(-1);
}
}
}
frames = this->spec.samples * 2;
status = ALSA_snd_pcm_hw_params_set_buffer_size_near(
this->hidden->pcm_handle, hwparams, &frames);
if ( status < 0 ) {
return(-1);
}
return ALSA_finalize_hardware(this, hwparams, override);
}
static int
ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
@ -653,11 +692,14 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
this->spec.freq = rate;
/* Set the buffer size, in samples */
status = ALSA_set_buffer_size(this, hwparams);
if (status < 0) {
return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
if ( ALSA_set_period_size(this, hwparams, 0) < 0 &&
ALSA_set_buffer_size(this, hwparams, 0) < 0 ) {
/* Failed to set desired buffer size, do the best you can... */
status = ALSA_set_period_size(this, hwparams, 1);
if (status < 0) {
return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
}
}
/* Set the software parameters */
snd_pcm_sw_params_alloca(&swparams);
status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
@ -695,11 +737,9 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
}
#if !SDL_ALSA_NON_BLOCKING
if (!iscapture) {
ALSA_snd_pcm_nonblock(pcm_handle, 0);
}
#endif
/* We're ready to rock and roll. :-) */
return 0;

25
Engine/lib/sdl/src/audio/android/SDL_androidaudio.c
View file

@ -57,9 +57,7 @@ ANDROIDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) { /* no "UNKNOWN" constant */
if ((test_format == AUDIO_U8) ||
(test_format == AUDIO_S16) ||
(test_format == AUDIO_F32)) {
if ((test_format == AUDIO_U8) || (test_format == AUDIO_S16LSB)) {
this->spec.format = test_format;
break;
}
@ -71,8 +69,25 @@ ANDROIDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
return SDL_SetError("No compatible audio format!");
}
if (Android_JNI_OpenAudioDevice(iscapture, &this->spec) < 0) {
return -1;
if (this->spec.channels > 1) {
this->spec.channels = 2;
} else {
this->spec.channels = 1;
}
if (this->spec.freq < 8000) {
this->spec.freq = 8000;
}
if (this->spec.freq > 48000) {
this->spec.freq = 48000;
}
/* TODO: pass in/return a (Java) device ID */
this->spec.samples = Android_JNI_OpenAudioDevice(iscapture, this->spec.freq, this->spec.format == AUDIO_U8 ? 0 : 1, this->spec.channels, this->spec.samples);
if (this->spec.samples == 0) {
/* Init failed? */
return SDL_SetError("Java-side initialization failed!");
}
SDL_CalculateAudioSpec(&this->spec);

2
Engine/lib/sdl/src/audio/arts/SDL_artsaudio.c
View file

@ -39,7 +39,7 @@
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#define SDL_NAME(X) X
#endif
#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC

8
Engine/lib/sdl/src/audio/coreaudio/SDL_coreaudio.h
View file

@ -45,14 +45,16 @@
struct SDL_PrivateAudioData
{
SDL_Thread *thread;
AudioQueueRef audioQueue;
int numAudioBuffers;
AudioQueueBufferRef *audioBuffer;
void *buffer;
UInt32 bufferOffset;
UInt32 bufferSize;
AudioStreamBasicDescription strdesc;
SDL_bool refill;
SDL_AudioStream *capturestream;
SDL_sem *ready_semaphore;
char *thread_error;
SDL_atomic_t shutdown;
#if MACOSX_COREAUDIO
AudioDeviceID deviceID;
#else

222
Engine/lib/sdl/src/audio/coreaudio/SDL_coreaudio.m
View file

@ -26,7 +26,6 @@
#include "SDL_audio.h"
#include "SDL_hints.h"
#include "SDL_timer.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "SDL_coreaudio.h"
@ -355,7 +354,7 @@ static BOOL update_audio_session(_THIS, SDL_bool open)
return NO;
}
if (open && (open_playback_devices + open_capture_devices) == 1) {
if (open_playback_devices + open_capture_devices == 1) {
if (![session setActive:YES error:&err]) {
NSString *desc = err.description;
SDL_SetError("Could not activate Audio Session: %s", desc.UTF8String);
@ -392,10 +391,10 @@ static BOOL update_audio_session(_THIS, SDL_bool open)
if (this->hidden->interruption_listener != NULL) {
SDLInterruptionListener *listener = nil;
listener = (SDLInterruptionListener *) CFBridgingRelease(this->hidden->interruption_listener);
[center removeObserver:listener];
@synchronized (listener) {
listener.device = NULL;
}
[center removeObserver:listener];
}
}
}
@ -410,27 +409,43 @@ static void
outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
SDL_assert(inBuffer->mAudioDataBytesCapacity == this->hidden->bufferSize);
SDL_memcpy(inBuffer->mAudioData, this->hidden->buffer, this->hidden->bufferSize);
SDL_memset(this->hidden->buffer, '\0', this->hidden->bufferSize); /* zero out in case we have to fill again without new data. */
inBuffer->mAudioDataByteSize = this->hidden->bufferSize;
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
this->hidden->refill = SDL_TRUE;
}
static Uint8 *
COREAUDIO_GetDeviceBuf(_THIS)
{
return this->hidden->buffer;
}
static void
COREAUDIO_WaitDevice(_THIS)
{
while (SDL_AtomicGet(&this->enabled) && !this->hidden->refill) {
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.10, 1);
if (SDL_AtomicGet(&this->hidden->shutdown)) {
return; /* don't do anything. */
}
this->hidden->refill = SDL_FALSE;
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
/* Supply silence if audio is not enabled or paused */
SDL_memset(inBuffer->mAudioData, this->spec.silence, inBuffer->mAudioDataBytesCapacity);
} else {
UInt32 remaining = inBuffer->mAudioDataBytesCapacity;
Uint8 *ptr = (Uint8 *) inBuffer->mAudioData;
while (remaining > 0) {
UInt32 len;
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
/* Generate the data */
SDL_LockMutex(this->mixer_lock);
(*this->callbackspec.callback)(this->callbackspec.userdata,
this->hidden->buffer, this->hidden->bufferSize);
SDL_UnlockMutex(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
len = this->hidden->bufferSize - this->hidden->bufferOffset;
if (len > remaining) {
len = remaining;
}
SDL_memcpy(ptr, (char *)this->hidden->buffer +
this->hidden->bufferOffset, len);
ptr = ptr + len;
remaining -= len;
this->hidden->bufferOffset += len;
}
}
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
inBuffer->mAudioDataByteSize = inBuffer->mAudioDataBytesCapacity;
}
static void
@ -439,46 +454,36 @@ inputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer
const AudioStreamPacketDescription *inPacketDescs )
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
if (SDL_AtomicGet(&this->enabled)) {
SDL_AudioStream *stream = this->hidden->capturestream;
if (SDL_AudioStreamPut(stream, inBuffer->mAudioData, inBuffer->mAudioDataByteSize) == -1) {
/* yikes, out of memory or something. I guess drop the buffer. Our WASAPI target kills the device in this case, though */
}
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
this->hidden->refill = SDL_TRUE;
}
}
static int
COREAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
SDL_AudioStream *stream = this->hidden->capturestream;
while (SDL_AtomicGet(&this->enabled)) {
const int avail = SDL_AudioStreamAvailable(stream);
if (avail > 0) {
const int cpy = SDL_min(buflen, avail);
SDL_AudioStreamGet(stream, buffer, cpy);
return cpy;
}
/* wait for more data, try again. */
while (SDL_AtomicGet(&this->enabled) && !this->hidden->refill) {
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.10, 1);
}
this->hidden->refill = SDL_FALSE;
if (SDL_AtomicGet(&this->shutdown)) {
return; /* don't do anything. */
}
return 0; /* not enabled, giving up. */
}
/* ignore unless we're active. */
if (!SDL_AtomicGet(&this->paused) && SDL_AtomicGet(&this->enabled) && !SDL_AtomicGet(&this->paused)) {
const Uint8 *ptr = (const Uint8 *) inBuffer->mAudioData;
UInt32 remaining = inBuffer->mAudioDataByteSize;
while (remaining > 0) {
UInt32 len = this->hidden->bufferSize - this->hidden->bufferOffset;
if (len > remaining) {
len = remaining;
}
static void
COREAUDIO_FlushCapture(_THIS)
{
while (CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0, 1) == kCFRunLoopRunHandledSource) {
/* spin. */
SDL_memcpy((char *)this->hidden->buffer + this->hidden->bufferOffset, ptr, len);
ptr += len;
remaining -= len;
this->hidden->bufferOffset += len;
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
SDL_LockMutex(this->mixer_lock);
(*this->callbackspec.callback)(this->callbackspec.userdata, this->hidden->buffer, this->hidden->bufferSize);
SDL_UnlockMutex(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
}
}
this->hidden->refill = SDL_FALSE;
SDL_AudioStreamClear(this->hidden->capturestream);
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
}
@ -536,16 +541,25 @@ COREAUDIO_CloseDevice(_THIS)
update_audio_session(this, SDL_FALSE);
#endif
/* if callback fires again, feed silence; don't call into the app. */
SDL_AtomicSet(&this->paused, 1);
if (this->hidden->audioQueue) {
AudioQueueDispose(this->hidden->audioQueue, 1);
}
if (this->hidden->capturestream) {
SDL_FreeAudioStream(this->hidden->capturestream);
if (this->hidden->thread) {
SDL_AtomicSet(&this->hidden->shutdown, 1);
SDL_WaitThread(this->hidden->thread, NULL);
}
if (this->hidden->ready_semaphore) {
SDL_DestroySemaphore(this->hidden->ready_semaphore);
}
/* AudioQueueDispose() frees the actual buffer objects. */
SDL_free(this->hidden->audioBuffer);
SDL_free(this->hidden->thread_error);
SDL_free(this->hidden->buffer);
SDL_free(this->hidden);
@ -611,8 +625,6 @@ prepare_device(_THIS, void *handle, int iscapture)
}
#endif
/* this all happens in the audio thread, since it needs a separate runloop. */
static int
prepare_audioqueue(_THIS)
{
@ -652,6 +664,19 @@ prepare_audioqueue(_THIS)
}
#endif
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate a sample buffer */
this->hidden->bufferSize = this->spec.size;
this->hidden->bufferOffset = iscapture ? 0 : this->hidden->bufferSize;
this->hidden->buffer = SDL_malloc(this->hidden->bufferSize);
if (this->hidden->buffer == NULL) {
SDL_OutOfMemory();
return 0;
}
/* Make sure we can feed the device a minimum amount of time */
double MINIMUM_AUDIO_BUFFER_TIME_MS = 15.0;
#if defined(__IPHONEOS__)
@ -666,7 +691,6 @@ prepare_audioqueue(_THIS)
numAudioBuffers = ((int)SDL_ceil(MINIMUM_AUDIO_BUFFER_TIME_MS / msecs) * 2);
}
this->hidden->numAudioBuffers = numAudioBuffers;
this->hidden->audioBuffer = SDL_calloc(1, sizeof (AudioQueueBufferRef) * numAudioBuffers);
if (this->hidden->audioBuffer == NULL) {
SDL_OutOfMemory();
@ -693,23 +717,29 @@ prepare_audioqueue(_THIS)
return 1;
}
static void
COREAUDIO_ThreadInit(_THIS)
static int
audioqueue_thread(void *arg)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
const int rc = prepare_audioqueue(this);
if (!rc) {
/* !!! FIXME: do this in RunAudio, and maybe block OpenDevice until ThreadInit finishes, too, to report an opening error */
SDL_OpenedAudioDeviceDisconnected(this); /* oh well. */
this->hidden->thread_error = SDL_strdup(SDL_GetError());
SDL_SemPost(this->hidden->ready_semaphore);
return 0;
}
}
static void
COREAUDIO_PrepareToClose(_THIS)
{
/* run long enough to queue some silence, so we know our actual audio
has been played */
CFRunLoopRunInMode(kCFRunLoopDefaultMode, (((this->spec.samples * 1000) / this->spec.freq) * 2) / 1000.0f, 0);
AudioQueueStop(this->hidden->audioQueue, 1);
/* init was successful, alert parent thread and start running... */
SDL_SemPost(this->hidden->ready_semaphore);
while (!SDL_AtomicGet(&this->hidden->shutdown)) {
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.10, 1);
}
if (!this->iscapture) { /* Drain off any pending playback. */
const CFTimeInterval secs = (((this->spec.size / (SDL_AUDIO_BITSIZE(this->spec.format) / 8)) / this->spec.channels) / ((CFTimeInterval) this->spec.freq)) * 2.0;
CFRunLoopRunInMode(kCFRunLoopDefaultMode, secs, 0);
}
return 0;
}
static int
@ -796,23 +826,28 @@ COREAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
#endif
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
if (iscapture) {
this->hidden->capturestream = SDL_NewAudioStream(this->spec.format, this->spec.channels, this->spec.freq, this->spec.format, this->spec.channels, this->spec.freq);
if (!this->hidden->capturestream) {
return -1; /* already set SDL_Error */
}
} else {
this->hidden->bufferSize = this->spec.size;
this->hidden->buffer = SDL_malloc(this->hidden->bufferSize);
if (this->hidden->buffer == NULL) {
return SDL_OutOfMemory();
}
/* This has to init in a new thread so it can get its own CFRunLoop. :/ */
SDL_AtomicSet(&this->hidden->shutdown, 0);
this->hidden->ready_semaphore = SDL_CreateSemaphore(0);
if (!this->hidden->ready_semaphore) {
return -1; /* oh well. */
}
return 0;
this->hidden->thread = SDL_CreateThreadInternal(audioqueue_thread, "AudioQueue thread", 512 * 1024, this);
if (!this->hidden->thread) {
return -1;
}
SDL_SemWait(this->hidden->ready_semaphore);
SDL_DestroySemaphore(this->hidden->ready_semaphore);
this->hidden->ready_semaphore = NULL;
if ((this->hidden->thread != NULL) && (this->hidden->thread_error != NULL)) {
SDL_SetError("%s", this->hidden->thread_error);
return -1;
}
return (this->hidden->thread != NULL) ? 0 : -1;
}
static void
@ -832,12 +867,6 @@ COREAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->OpenDevice = COREAUDIO_OpenDevice;
impl->CloseDevice = COREAUDIO_CloseDevice;
impl->Deinitialize = COREAUDIO_Deinitialize;
impl->ThreadInit = COREAUDIO_ThreadInit;
impl->WaitDevice = COREAUDIO_WaitDevice;
impl->GetDeviceBuf = COREAUDIO_GetDeviceBuf;
impl->PrepareToClose = COREAUDIO_PrepareToClose;
impl->CaptureFromDevice = COREAUDIO_CaptureFromDevice;
impl->FlushCapture = COREAUDIO_FlushCapture;
#if MACOSX_COREAUDIO
impl->DetectDevices = COREAUDIO_DetectDevices;
@ -847,6 +876,7 @@ COREAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->OnlyHasDefaultCaptureDevice = 1;
#endif
impl->ProvidesOwnCallbackThread = 1;
impl->HasCaptureSupport = 1;
return 1; /* this audio target is available. */

6
Engine/lib/sdl/src/audio/directsound/SDL_directsound.c
View file

@ -477,8 +477,8 @@ DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_bool tried_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
LPGUID guid = (LPGUID) handle;
DWORD bufsize;
DWORD bufsize;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
@ -526,7 +526,7 @@ DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
(int) (DSBSIZE_MAX / numchunks));
} else {
int rc;
WAVEFORMATEX wfmt;
WAVEFORMATEX wfmt;
SDL_zero(wfmt);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
wfmt.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;

35
Engine/lib/sdl/src/audio/jack/SDL_jackaudio.c
View file

@ -44,9 +44,7 @@ static const char ** (*JACK_jack_get_ports) (jack_client_t *, const char *, cons
static jack_nframes_t (*JACK_jack_get_sample_rate) (jack_client_t *);
static jack_nframes_t (*JACK_jack_get_buffer_size) (jack_client_t *);
static jack_port_t * (*JACK_jack_port_register) (jack_client_t *, const char *, const char *, unsigned long, unsigned long);
static jack_port_t * (*JACK_jack_port_by_name) (jack_client_t *, const char *);
static const char * (*JACK_jack_port_name) (const jack_port_t *);
static const char * (*JACK_jack_port_type) (const jack_port_t *);
static int (*JACK_jack_connect) (jack_client_t *, const char *, const char *);
static int (*JACK_jack_set_process_callback) (jack_client_t *, JackProcessCallback, void *);
@ -137,9 +135,7 @@ load_jack_syms(void)
SDL_JACK_SYM(jack_get_sample_rate);
SDL_JACK_SYM(jack_get_buffer_size);
SDL_JACK_SYM(jack_port_register);
SDL_JACK_SYM(jack_port_by_name);
SDL_JACK_SYM(jack_port_name);
SDL_JACK_SYM(jack_port_type);
SDL_JACK_SYM(jack_connect);
SDL_JACK_SYM(jack_set_process_callback);
return 0;
@ -277,6 +273,10 @@ JACK_CloseDevice(_THIS)
SDL_DestroySemaphore(this->hidden->iosem);
}
if (this->hidden->devports) {
JACK_jack_free(this->hidden->devports);
}
SDL_free(this->hidden->iobuffer);
}
@ -292,11 +292,9 @@ JACK_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
const JackProcessCallback callback = iscapture ? jackProcessCaptureCallback : jackProcessPlaybackCallback;
const char *sdlportstr = iscapture ? "input" : "output";
const char **devports = NULL;
int *audio_ports;
jack_client_t *client = NULL;
jack_status_t status;
int channels = 0;
int ports = 0;
int i;
/* Initialize all variables that we clean on shutdown */
@ -313,30 +311,15 @@ JACK_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
devports = JACK_jack_get_ports(client, NULL, NULL, JackPortIsPhysical | sysportflags);
this->hidden->devports = devports;
if (!devports || !devports[0]) {
return SDL_SetError("No physical JACK ports available");
}
while (devports[++ports]) {
while (devports[++channels]) {
/* spin to count devports */
}
/* Filter out non-audio ports */
audio_ports = SDL_calloc(ports, sizeof *audio_ports);
for (i = 0; i < ports; i++) {
const jack_port_t *dport = JACK_jack_port_by_name(client, devports[i]);
const char *type = JACK_jack_port_type(dport);
const int len = SDL_strlen(type);
/* See if type ends with "audio" */
if (len >= 5 && !SDL_memcmp(type+len-5, "audio", 5)) {
audio_ports[channels++] = i;
}
}
if (channels == 0) {
return SDL_SetError("No physical JACK ports available");
}
/* !!! FIXME: docs say about buffer size: "This size may change, clients that depend on it must register a bufsize_callback so they will be notified if it does." */
/* Jack pretty much demands what it wants. */
@ -385,16 +368,16 @@ JACK_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* once activated, we can connect all the ports. */
for (i = 0; i < channels; i++) {
const char *sdlport = JACK_jack_port_name(this->hidden->sdlports[i]);
const char *srcport = iscapture ? devports[audio_ports[i]] : sdlport;
const char *dstport = iscapture ? sdlport : devports[audio_ports[i]];
const char *srcport = iscapture ? devports[i] : sdlport;
const char *dstport = iscapture ? sdlport : devports[i];
if (JACK_jack_connect(client, srcport, dstport) != 0) {
return SDL_SetError("Couldn't connect JACK ports: %s => %s", srcport, dstport);
}
}
/* don't need these anymore. */
this->hidden->devports = NULL;
JACK_jack_free(devports);
SDL_free(audio_ports);
/* We're ready to rock and roll. :-) */
return 0;

1
Engine/lib/sdl/src/audio/jack/SDL_jackaudio.h
View file

@ -33,6 +33,7 @@ struct SDL_PrivateAudioData
jack_client_t *client;
SDL_sem *iosem;
float *iobuffer;
const char **devports;
jack_port_t **sdlports;
};

2
Engine/lib/sdl/src/audio/pulseaudio/SDL_pulseaudio.c
View file

@ -109,7 +109,7 @@ static pa_operation * (*PULSEAUDIO_pa_stream_drain) (pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_peek) (pa_stream *, const void **, size_t *);
static int (*PULSEAUDIO_pa_stream_drop) (pa_stream *);
static pa_operation * (*PULSEAUDIO_pa_stream_flush) (pa_stream *,
static pa_operation * (*PULSEAUDIO_pa_stream_flush) (pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_disconnect) (pa_stream *);
static void (*PULSEAUDIO_pa_stream_unref) (pa_stream *);

6
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi.c
View file

@ -725,12 +725,6 @@ WASAPI_ThreadDeinit(_THIS)
WASAPI_PlatformThreadDeinit(this);
}
void
WASAPI_BeginLoopIteration(_THIS)
{
/* no-op. */
}
static void
WASAPI_Deinitialize(void)
{

68
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi_win32.c
View file

@ -351,42 +351,10 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
}
typedef struct
{
LPWSTR devid;
char *devname;
} EndpointItem;
static int sort_endpoints(const void *_a, const void *_b)
{
LPWSTR a = ((const EndpointItem *) _a)->devid;
LPWSTR b = ((const EndpointItem *) _b)->devid;
if (!a && b) {
return -1;
} else if (a && !b) {
return 1;
}
while (SDL_TRUE) {
if (*a < *b) {
return -1;
} else if (*a > *b) {
return 1;
} else if (*a == 0) {
break;
}
a++;
b++;
}
return 0;
}
static void
WASAPI_EnumerateEndpointsForFlow(const SDL_bool iscapture)
{
IMMDeviceCollection *collection = NULL;
EndpointItem *items;
UINT i, total;
/* Note that WASAPI separates "adapter devices" from "audio endpoint devices"
@ -401,36 +369,22 @@ WASAPI_EnumerateEndpointsForFlow(const SDL_bool iscapture)
return;
}
items = (EndpointItem *) SDL_calloc(total, sizeof (EndpointItem));
if (!items) {
return; /* oh well. */
}
for (i = 0; i < total; i++) {
EndpointItem *item = items + i;
IMMDevice *device = NULL;
if (SUCCEEDED(IMMDeviceCollection_Item(collection, i, &device))) {
if (SUCCEEDED(IMMDevice_GetId(device, &item->devid))) {
item->devname = GetWasapiDeviceName(device);
LPWSTR devid = NULL;
if (SUCCEEDED(IMMDevice_GetId(device, &devid))) {
char *devname = GetWasapiDeviceName(device);
if (devname) {
WASAPI_AddDevice(iscapture, devname, devid);
SDL_free(devname);
}
CoTaskMemFree(devid);
}
IMMDevice_Release(device);
}
}
/* sort the list of devices by their guid so list is consistent between runs */
SDL_qsort(items, total, sizeof (*items), sort_endpoints);
/* Send the sorted list on to the SDL's higher level. */
for (i = 0; i < total; i++) {
EndpointItem *item = items + i;
if ((item->devid) && (item->devname)) {
WASAPI_AddDevice(iscapture, item->devname, item->devid);
}
SDL_free(item->devname);
CoTaskMemFree(item->devid);
}
SDL_free(items);
IMMDeviceCollection_Release(collection);
}
@ -451,6 +405,12 @@ WASAPI_PlatformDeleteActivationHandler(void *handler)
SDL_assert(!"This function should have only been called on WinRT.");
}
void
WASAPI_BeginLoopIteration(_THIS)
{
/* no-op. */
}
#endif /* SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__) */
/* vi: set ts=4 sw=4 expandtab: */

65
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi_winrt.cpp
View file

@ -185,9 +185,20 @@ struct SDL_WasapiActivationHandler : public RuntimeClass< RuntimeClassFlags< Cla
HRESULT
SDL_WasapiActivationHandler::ActivateCompleted(IActivateAudioInterfaceAsyncOperation *async)
{
// Just set a flag, since we're probably in a different thread. We'll pick it up and init everything on our own thread to prevent races.
SDL_AtomicSet(&device->hidden->just_activated, 1);
HRESULT result = S_OK;
IUnknown *iunknown = nullptr;
const HRESULT ret = async->GetActivateResult(&result, &iunknown);
if (SUCCEEDED(ret) && SUCCEEDED(result)) {
iunknown->QueryInterface(IID_PPV_ARGS(&device->hidden->client));
if (device->hidden->client) {
// Just set a flag, since we're probably in a different thread. We'll pick it up and init everything on our own thread to prevent races.
SDL_AtomicSet(&device->hidden->just_activated, 1);
}
}
WASAPI_UnrefDevice(device);
return S_OK;
}
@ -225,49 +236,29 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
IActivateAudioInterfaceAsyncOperation *async = nullptr;
const HRESULT ret = ActivateAudioInterfaceAsync(devid, __uuidof(IAudioClient), nullptr, handler.Get(), &async);
if (FAILED(ret) || async == nullptr) {
if (async != nullptr) {
async->Release();
}
if (async != nullptr) {
async->Release();
}
if (FAILED(ret)) {
handler.Get()->Release();
WASAPI_UnrefDevice(_this);
return WIN_SetErrorFromHRESULT("WASAPI can't activate requested audio endpoint", ret);
}
/* Spin until the async operation is complete.
* If we don't PrepDevice before leaving this function, the bug list gets LONG:
* - device.spec is not filled with the correct information
* - The 'obtained' spec will be wrong for ALLOW_CHANGE properties
* - SDL_AudioStreams will/will not be allocated at the right time
* - SDL_assert(device->callbackspec.size == device->spec.size) will fail
* - When the assert is ignored, skipping or a buffer overflow will occur
*/
while (!SDL_AtomicCAS(&_this->hidden->just_activated, 1, 0)) {
SDL_Delay(1);
}
HRESULT activateRes = S_OK;
IUnknown *iunknown = nullptr;
const HRESULT getActivateRes = async->GetActivateResult(&activateRes, &iunknown);
async->Release();
if (FAILED(getActivateRes)) {
return WIN_SetErrorFromHRESULT("Failed to get WASAPI activate result", getActivateRes);
} else if (FAILED(activateRes)) {
return WIN_SetErrorFromHRESULT("Failed to activate WASAPI device", activateRes);
}
iunknown->QueryInterface(IID_PPV_ARGS(&_this->hidden->client));
if (!_this->hidden->client) {
return SDL_SetError("Failed to query WASAPI client interface");
}
if (WASAPI_PrepDevice(_this, isrecovery) == -1) {
return -1;
}
return 0;
}
void
WASAPI_BeginLoopIteration(_THIS)
{
if (SDL_AtomicCAS(&_this->hidden->just_activated, 1, 0)) {
if (WASAPI_PrepDevice(_this, SDL_TRUE) == -1) {
SDL_OpenedAudioDeviceDisconnected(_this);
}
}
}
void
WASAPI_PlatformThreadInit(_THIS)
{

5
Engine/lib/sdl/src/audio/winmm/SDL_winmm.c
View file

@ -78,7 +78,7 @@ static void DetectWave##typ##Devs(void) { \
capstyp##2W caps; \
UINT i; \
for (i = 0; i < devcount; i++) { \
if (wave##typ##GetDevCaps(i,(LP##capstyp##W)&caps,sizeof(caps))==MMSYSERR_NOERROR) { \
if (wave##typ##GetDevCaps(i,(LP##capstyp##W)&caps,sizeof(caps))==MMSYSERR_NOERROR) { \
char *name = WIN_LookupAudioDeviceName(caps.szPname,&caps.NameGuid); \
if (name != NULL) { \
SDL_AddAudioDevice((int) iscapture, name, (void *) ((size_t) i+1)); \
@ -375,7 +375,8 @@ WINMM_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
#endif
/* Create the audio buffer semaphore */
this->hidden->audio_sem = CreateSemaphore(NULL, iscapture ? 0 : NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
this->hidden->audio_sem =
CreateSemaphore(NULL, iscapture ? 0 : NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
if (this->hidden->audio_sem == NULL) {
return SDL_SetError("Couldn't create semaphore");
}