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Update to SDL2.0.10

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This commit is contained in:
Areloch 2019-08-19 23:30:35 -05:00
parent 600859bd63
commit c932bda8dd
915 changed files with 116675 additions and 21754 deletions
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145
Engine/lib/sdl/src/audio/SDL_audio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -89,6 +89,9 @@ static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_FUSIONSOUND
&FUSIONSOUND_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_OPENSLES
&openslES_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_ANDROID
&ANDROIDAUDIO_bootstrap,
#endif
@ -245,12 +248,6 @@ SDL_AudioPlayDevice_Default(_THIS)
{ /* no-op. */
}
static int
SDL_AudioGetPendingBytes_Default(_THIS)
{
return 0;
}
static Uint8 *
SDL_AudioGetDeviceBuf_Default(_THIS)
{
@ -358,7 +355,6 @@ finish_audio_entry_points_init(void)
FILL_STUB(BeginLoopIteration);
FILL_STUB(WaitDevice);
FILL_STUB(PlayDevice);
FILL_STUB(GetPendingBytes);
FILL_STUB(GetDeviceBuf);
FILL_STUB(CaptureFromDevice);
FILL_STUB(FlushCapture);
@ -378,21 +374,57 @@ static int
add_audio_device(const char *name, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
{
int retval = -1;
const size_t size = sizeof (SDL_AudioDeviceItem) + SDL_strlen(name) + 1;
SDL_AudioDeviceItem *item = (SDL_AudioDeviceItem *) SDL_malloc(size);
if (item == NULL) {
return -1;
}
SDL_AudioDeviceItem *item;
const SDL_AudioDeviceItem *i;
int dupenum = 0;
SDL_assert(handle != NULL); /* we reserve NULL, audio backends can't use it. */
SDL_assert(name != NULL);
item = (SDL_AudioDeviceItem *) SDL_malloc(sizeof (SDL_AudioDeviceItem));
if (!item) {
return SDL_OutOfMemory();
}
item->original_name = SDL_strdup(name);
if (!item->original_name) {
SDL_free(item);
return SDL_OutOfMemory();
}
item->dupenum = 0;
item->name = item->original_name;
item->handle = handle;
SDL_strlcpy(item->name, name, size - sizeof (SDL_AudioDeviceItem));
SDL_LockMutex(current_audio.detectionLock);
for (i = *devices; i != NULL; i = i->next) {
if (SDL_strcmp(name, i->original_name) == 0) {
dupenum = i->dupenum + 1;
break; /* stop at the highest-numbered dupe. */
}
}
if (dupenum) {
const size_t len = SDL_strlen(name) + 16;
char *replacement = (char *) SDL_malloc(len);
if (!replacement) {
SDL_UnlockMutex(current_audio.detectionLock);
SDL_free(item->original_name);
SDL_free(item);
SDL_OutOfMemory();
return -1;
}
SDL_snprintf(replacement, len, "%s (%d)", name, dupenum + 1);
item->dupenum = dupenum;
item->name = replacement;
}
item->next = *devices;
*devices = item;
retval = (*devCount)++;
retval = (*devCount)++; /* !!! FIXME: this should be an atomic increment */
SDL_UnlockMutex(current_audio.detectionLock);
return retval;
@ -420,6 +452,11 @@ 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;
@ -451,7 +488,11 @@ void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
SDL_assert(get_audio_device(device->id) == device);
if (!SDL_AtomicGet(&device->enabled)) {
return;
return; /* don't report disconnects more than once. */
}
if (SDL_AtomicGet(&device->shutdown)) {
return; /* don't report disconnect if we're trying to close device. */
}
/* Ends the audio callback and mark the device as STOPPED, but the
@ -606,11 +647,9 @@ SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
}
/* Nothing to do unless we're set up for queueing. */
if (device->callbackspec.callback == SDL_BufferQueueDrainCallback) {
current_audio.impl.LockDevice(device);
retval = ((Uint32) SDL_CountDataQueue(device->buffer_queue)) + current_audio.impl.GetPendingBytes(device);
current_audio.impl.UnlockDevice(device);
} else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
if (device->callbackspec.callback == SDL_BufferQueueDrainCallback ||
device->callbackspec.callback == SDL_BufferQueueFillCallback)
{
current_audio.impl.LockDevice(device);
retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
current_audio.impl.UnlockDevice(device);
@ -650,8 +689,16 @@ SDL_RunAudio(void *devicep)
SDL_assert(!device->iscapture);
#if SDL_AUDIO_DRIVER_ANDROID
{
/* Set thread priority to THREAD_PRIORITY_AUDIO */
extern void Android_JNI_AudioSetThreadPriority(int, int);
Android_JNI_AudioSetThreadPriority(device->iscapture, device->id);
}
#else
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_TIME_CRITICAL);
#endif
/* Perform any thread setup */
device->threadid = SDL_ThreadID();
@ -747,8 +794,16 @@ SDL_CaptureAudio(void *devicep)
SDL_assert(device->iscapture);
#if SDL_AUDIO_DRIVER_ANDROID
{
/* Set thread priority to THREAD_PRIORITY_AUDIO */
extern void Android_JNI_AudioSetThreadPriority(int, int);
Android_JNI_AudioSetThreadPriority(device->iscapture, device->id);
}
#else
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
#endif
/* Perform any thread setup */
device->threadid = SDL_ThreadID();
@ -971,6 +1026,11 @@ 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;
@ -997,7 +1057,6 @@ 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;
@ -1054,16 +1113,14 @@ close_audio_device(SDL_AudioDevice * device)
return;
}
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;
}
}
/* make sure the device is paused before we do anything else, so the
audio callback definitely won't fire again. */
current_audio.impl.LockDevice(device);
SDL_AtomicSet(&device->paused, 1);
SDL_AtomicSet(&device->shutdown, 1);
SDL_AtomicSet(&device->enabled, 0);
current_audio.impl.UnlockDevice(device);
if (device->thread != NULL) {
SDL_WaitThread(device->thread, NULL);
}
@ -1074,6 +1131,14 @@ 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);
}
@ -1118,8 +1183,9 @@ prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
}
case 1: /* Mono */
case 2: /* Stereo */
case 4: /* surround */
case 6: /* surround with center and lfe */
case 4: /* Quadrophonic */
case 6: /* 5.1 surround */
case 8: /* 7.1 surround */
break;
default:
SDL_SetError("Unsupported number of audio channels.");
@ -1312,15 +1378,12 @@ 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) {
build_stream = SDL_TRUE;
if (allowed_changes & SDL_AUDIO_ALLOW_SAMPLES_CHANGE) {
obtained->samples = device->spec.samples;
} else {
build_stream = SDL_TRUE;
}
}
SDL_CalculateAudioSpec(obtained); /* recalc after possible changes. */

2
Engine/lib/sdl/src/audio/SDL_audio_c.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

26
Engine/lib/sdl/src/audio/SDL_audiocvt.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -718,13 +718,19 @@ SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format
/* !!! FIXME: remove this if we can get the resampler to work in-place again. */
float *dst = (float *) (cvt->buf + srclen);
const int dstlen = (cvt->len * cvt->len_mult) - srclen;
const int paddingsamples = (ResamplerPadding(inrate, outrate) * chans);
const int requestedpadding = ResamplerPadding(inrate, outrate);
int paddingsamples;
float *padding;
if (requestedpadding < SDL_MAX_SINT32 / chans) {
paddingsamples = requestedpadding * chans;
} else {
paddingsamples = 0;
}
SDL_assert(format == AUDIO_F32SYS);
/* we keep no streaming state here, so pad with silence on both ends. */
padding = (float *) SDL_calloc(paddingsamples, sizeof (float));
padding = (float *) SDL_calloc(paddingsamples ? paddingsamples : 1, sizeof (float));
if (!padding) {
SDL_OutOfMemory();
return;
@ -889,10 +895,14 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
return SDL_SetError("Invalid source channels");
} else if (!SDL_SupportedChannelCount(dst_channels)) {
return SDL_SetError("Invalid destination channels");
} else if (src_rate == 0) {
return SDL_SetError("Source rate is zero");
} else if (dst_rate == 0) {
return SDL_SetError("Destination rate is zero");
} else if (src_rate <= 0) {
return SDL_SetError("Source rate is equal to or less than zero");
} else if (dst_rate <= 0) {
return SDL_SetError("Destination rate is equal to or less than zero");
} else if (src_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
return SDL_SetError("Source rate is too high");
} else if (dst_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
return SDL_SetError("Destination rate is too high");
}
#if DEBUG_CONVERT
@ -1291,7 +1301,7 @@ 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, sizeof (float));
retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples ? retval->resampler_padding_samples : 1, sizeof (float));
if (retval->resampler_padding == NULL) {
SDL_FreeAudioStream(retval);

2
Engine/lib/sdl/src/audio/SDL_audiodev.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

8
Engine/lib/sdl/src/audio/SDL_audiodev_c.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -18,6 +18,10 @@
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"
@ -35,4 +39,6 @@
extern void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int));
#endif /* SDL_audiodev_c_h_ */
/* vi: set ts=4 sw=4 expandtab: */

741
Engine/lib/sdl/src/audio/SDL_audiotypecvt.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -25,8 +25,9 @@
#include "SDL_cpuinfo.h"
#include "SDL_assert.h"
/* !!! FIXME: write NEON code. */
#define HAVE_NEON_INTRINSICS 0
#ifdef __ARM_NEON
#define HAVE_NEON_INTRINSICS 1
#endif
#ifdef __SSE2__
#define HAVE_SSE2_INTRINSICS 1
@ -62,7 +63,7 @@ SDL_AudioFilter SDL_Convert_F32_to_S32 = NULL;
#define DIVBY128 0.0078125f
#define DIVBY32768 0.000030517578125f
#define DIVBY2147483648 0.00000000046566128730773926
#define DIVBY8388607 0.00000011920930376163766f
#if NEED_SCALAR_CONVERTER_FALLBACKS
@ -152,7 +153,7 @@ 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) (((double) *src) * DIVBY2147483648);
*dst = ((float) (*src>>8)) * DIVBY8388607;
}
if (cvt->filters[++cvt->filter_index]) {
@ -171,10 +172,10 @@ 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 = -127;
} else if (sample <= -1.0f) {
*dst = -128;
} else {
*dst = (Sint8)(sample * 127.0f);
}
@ -197,9 +198,9 @@ 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);
@ -223,10 +224,10 @@ 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 = -32767;
} else if (sample <= -1.0f) {
*dst = -32768;
} else {
*dst = (Sint16)(sample * 32767.0f);
}
@ -249,9 +250,9 @@ 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 = 65534;
} else if (sample < -1.0f) {
if (sample >= 1.0f) {
*dst = 65535;
} else if (sample <= -1.0f) {
*dst = 0;
} else {
*dst = (Uint16)((sample + 1.0f) * 32767.0f);
@ -275,12 +276,12 @@ 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 = -2147483647;
} else if (sample <= -1.0f) {
*dst = (Sint32) -2147483648LL;
} else {
*dst = (Sint32)((double)sample * 2147483647.0);
*dst = ((Sint32)(sample * 8388607.0f)) << 8;
}
}
@ -509,16 +510,6 @@ 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)
{
@ -530,23 +521,19 @@ 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) (((double) *src) * DIVBY2147483648);
*dst = ((float) (*src>>8)) * DIVBY8388607;
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
SDL_assert(!i || ((((size_t) src) & 15) == 0));
{
/* 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 __m128d divby2147483648 = _mm_set1_pd(DIVBY2147483648);
const __m128 divby8388607 = _mm_set1_ps(DIVBY8388607);
const __m128i *mmsrc = (const __m128i *) src;
while (i >= 4) { /* 4 * sint32 */
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)))));
/* 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));
i -= 4; mmsrc++; dst += 4;
}
src = (const Sint32 *) mmsrc;
@ -554,7 +541,7 @@ SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (float) (((double) *src) * DIVBY2147483648);
*dst = ((float) (*src>>8)) * DIVBY8388607;
i--; src++; dst++;
}
@ -574,7 +561,14 @@ 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) {
*dst = (Sint8) (*src * 127.0f);
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));
@ -582,13 +576,15 @@ 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_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 */
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 */
_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++;
}
@ -597,7 +593,14 @@ SDL_Convert_F32_to_S8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Sint8) (*src * 127.0f);
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++;
}
@ -618,7 +621,14 @@ 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) {
*dst = (Uint8) ((*src + 1.0f) * 127.0f);
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));
@ -626,14 +636,15 @@ 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 add1 = _mm_set1_ps(1.0f);
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_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 */
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 */
_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++;
}
@ -642,7 +653,14 @@ SDL_Convert_F32_to_U8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Uint8) ((*src + 1.0f) * 127.0f);
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++;
}
@ -663,7 +681,14 @@ 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) {
*dst = (Sint16) (*src * 32767.0f);
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));
@ -671,11 +696,13 @@ 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_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 */
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 */
_mm_store_si128(mmdst, _mm_packs_epi32(ints1, ints2)); /* pack to sint16, store out. */
i -= 8; src += 8; mmdst++;
}
@ -684,7 +711,14 @@ SDL_Convert_F32_to_S16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Sint16) (*src * 32767.0f);
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++;
}
@ -705,7 +739,14 @@ 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) {
*dst = (Uint16) ((*src + 1.0f) * 32767.0f);
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));
@ -722,10 +763,12 @@ 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_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 */
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 */
_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++;
}
@ -734,7 +777,14 @@ SDL_Convert_F32_to_U16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Uint16) ((*src + 1.0f) * 32767.0f);
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++;
}
@ -755,7 +805,14 @@ 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) {
*dst = (Sint32) (((double) *src) * 2147483647.0);
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;
}
}
SDL_assert(!i || ((((size_t) dst) & 15) == 0));
@ -763,14 +820,12 @@ 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 __m128d mulby2147483647 = _mm_set1_pd(2147483647.0);
const __m128 one = _mm_set1_ps(1.0f);
const __m128 negone = _mm_set1_ps(-1.0f);
const __m128 mulby8388607 = _mm_set1_ps(8388607.0f);
__m128i *mmdst = (__m128i *) dst;
while (i >= 4) { /* 4 * float32 */
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)));
_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 */
i -= 4; src += 4; mmdst++;
}
dst = (Sint32 *) mmdst;
@ -778,7 +833,14 @@ SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format)
/* Finish off any leftovers with scalar operations. */
while (i) {
*dst = (Sint32) (((double) *src) * 2147483647.0);
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;
}
i--; src++; dst++;
}
@ -789,6 +851,538 @@ 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 negone = 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, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16hi))), divby128));
vst1q_f32(dst+4, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16hi))), divby128));
vst1q_f32(dst+8, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16lo))), divby128));
vst1q_f32(dst+12, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16lo))), divby128));
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 negone = 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, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uints))), divby32768));
vst1q_f32(dst+4, vmlaq_f32(negone, 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));
/* 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 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 = (-2147483647) - 1;
} 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 = (-2147483647) - 1;
} 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;
@ -817,6 +1411,13 @@ 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

2
Engine/lib/sdl/src/audio/SDL_mixer.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

8
Engine/lib/sdl/src/audio/SDL_sysaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -71,7 +71,6 @@ typedef struct SDL_AudioDriverImpl
void (*BeginLoopIteration)(_THIS); /* Called by audio thread at top of loop */
void (*WaitDevice) (_THIS);
void (*PlayDevice) (_THIS);
int (*GetPendingBytes) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
int (*CaptureFromDevice) (_THIS, void *buffer, int buflen);
void (*FlushCapture) (_THIS);
@ -98,8 +97,10 @@ 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;
@ -202,6 +203,7 @@ extern AudioBootStrap COREAUDIO_bootstrap;
extern AudioBootStrap DISKAUDIO_bootstrap;
extern AudioBootStrap DUMMYAUDIO_bootstrap;
extern AudioBootStrap FUSIONSOUND_bootstrap;
extern AudioBootStrap openslES_bootstrap;
extern AudioBootStrap ANDROIDAUDIO_bootstrap;
extern AudioBootStrap PSPAUDIO_bootstrap;
extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;

2515
Engine/lib/sdl/src/audio/SDL_wave.c
View file

File diff suppressed because it is too large Load diff

136
Engine/lib/sdl/src/audio/SDL_wave.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -20,11 +20,12 @@
*/
#include "../SDL_internal.h"
/* WAVE files are little-endian */
/* RIFF WAVE files are little-endian */
/*******************************************/
/* Define values for Microsoft WAVE format */
/*******************************************/
/* FOURCC */
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
@ -33,45 +34,116 @@
#define JUNK 0x4B4E554A /* "JUNK" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
/* Format tags */
#define UNKNOWN_CODE 0x0000
#define PCM_CODE 0x0001
#define MS_ADPCM_CODE 0x0002
#define IEEE_FLOAT_CODE 0x0003
#define ALAW_CODE 0x0006
#define MULAW_CODE 0x0007
#define IMA_ADPCM_CODE 0x0011
#define MP3_CODE 0x0055
#define MPEG_CODE 0x0050
#define MPEGLAYER3_CODE 0x0055
#define EXTENSIBLE_CODE 0xFFFE
#define WAVE_MONO 1
#define WAVE_STEREO 2
/* Normally, these three chunks come consecutively in a WAVE file */
typedef struct WaveFMT
/* Stores the WAVE format information. */
typedef struct WaveFormat
{
/* Not saved in the chunk we read:
Uint32 FMTchunk;
Uint32 fmtlen;
*/
Uint16 encoding;
Uint16 channels; /* 1 = mono, 2 = stereo */
Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */
Uint32 byterate; /* Average bytes per second */
Uint16 blockalign; /* Bytes per sample block */
Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */
} WaveFMT;
Uint16 formattag; /* Raw value of the first field in the fmt chunk data. */
Uint16 encoding; /* Actual encoding, possibly from the extensible header. */
Uint16 channels; /* Number of channels. */
Uint32 frequency; /* Sampling rate in Hz. */
Uint32 byterate; /* Average bytes per second. */
Uint16 blockalign; /* Bytes per block. */
Uint16 bitspersample; /* Currently supported are 8, 16, 24, 32, and 4 for ADPCM. */
/* The general chunk found in the WAVE file */
typedef struct Chunk
{
Uint32 magic;
Uint32 length;
Uint8 *data;
} Chunk;
/* Extra information size. Number of extra bytes starting at byte 18 in the
* fmt chunk data. This is at least 22 for the extensible header.
*/
Uint16 extsize;
typedef struct WaveExtensibleFMT
{
WaveFMT format;
Uint16 size;
Uint16 validbits;
/* Extensible WAVE header fields */
Uint16 validsamplebits;
Uint32 samplesperblock; /* For compressed formats. Can be zero. Actually 16 bits in the header. */
Uint32 channelmask;
Uint8 subformat[16]; /* a GUID. */
} WaveExtensibleFMT;
Uint8 subformat[16]; /* A format GUID. */
} WaveFormat;
/* Stores information on the fact chunk. */
typedef struct WaveFact {
/* Represents the state of the fact chunk in the WAVE file.
* Set to -1 if the fact chunk is invalid.
* Set to 0 if the fact chunk is not present
* Set to 1 if the fact chunk is present and valid.
* Set to 2 if samplelength is going to be used as the number of sample frames.
*/
Sint32 status;
/* Version 1 of the RIFF specification calls the field in the fact chunk
* dwFileSize. The Standards Update then calls it dwSampleLength and specifies
* that it is 'the length of the data in samples'. WAVE files from Windows
* with this chunk have it set to the samples per channel (sample frames).
* This is useful to truncate compressed audio to a specific sample count
* because a compressed block is usually decoded to a fixed number of
* sample frames.
*/
Uint32 samplelength; /* Raw sample length value from the fact chunk. */
} WaveFact;
/* Generic struct for the chunks in the WAVE file. */
typedef struct WaveChunk
{
Uint32 fourcc; /* FOURCC of the chunk. */
Uint32 length; /* Size of the chunk data. */
Sint64 position; /* Position of the data in the stream. */
Uint8 *data; /* When allocated, this points to the chunk data. length is used for the malloc size. */
size_t size; /* Number of bytes in data that could be read from the stream. Can be smaller than length. */
} WaveChunk;
/* Controls how the size of the RIFF chunk affects the loading of a WAVE file. */
typedef enum WaveRiffSizeHint {
RiffSizeNoHint,
RiffSizeForce,
RiffSizeIgnoreZero,
RiffSizeIgnore,
RiffSizeMaximum
} WaveRiffSizeHint;
/* Controls how a truncated WAVE file is handled. */
typedef enum WaveTruncationHint {
TruncNoHint,
TruncVeryStrict,
TruncStrict,
TruncDropFrame,
TruncDropBlock
} WaveTruncationHint;
/* Controls how the fact chunk affects the loading of a WAVE file. */
typedef enum WaveFactChunkHint {
FactNoHint,
FactTruncate,
FactStrict,
FactIgnoreZero,
FactIgnore
} WaveFactChunkHint;
typedef struct WaveFile
{
WaveChunk chunk;
WaveFormat format;
WaveFact fact;
/* Number of sample frames that will be decoded. Calculated either with the
* size of the data chunk or, if the appropriate hint is enabled, with the
* sample length value from the fact chunk.
*/
Sint64 sampleframes;
void *decoderdata; /* Some decoders require extra data for a state. */
WaveRiffSizeHint riffhint;
WaveTruncationHint trunchint;
WaveFactChunkHint facthint;
} WaveFile;
/* vi: set ts=4 sw=4 expandtab: */

173
Engine/lib/sdl/src/audio/alsa/SDL_alsa_audio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -22,6 +22,10 @@
#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>
@ -68,7 +72,9 @@ static int (*ALSA_snd_pcm_hw_params_set_period_size_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_get_period_size)
(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_near)
static int (*ALSA_snd_pcm_hw_params_set_periods_min)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_first)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_get_periods)
(const snd_pcm_hw_params_t *, unsigned int *, int *);
@ -90,6 +96,7 @@ 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);
@ -143,7 +150,8 @@ load_alsa_syms(void)
SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);
SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);
SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near);
SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_min);
SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_first);
SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);
SDL_ALSA_SYM(snd_pcm_hw_params_set_buffer_size_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size);
@ -158,6 +166,7 @@ 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);
@ -243,7 +252,24 @@ get_audio_device(void *handle, const int channels)
static void
ALSA_WaitDevice(_THIS)
{
/* We're in blocking mode, so there's nothing to do here */
#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
}
@ -422,7 +448,7 @@ 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;
@ -435,10 +461,38 @@ ALSA_CloseDevice(_THIS)
}
static int
ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
{
int status;
snd_pcm_uframes_t bufsize;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t persize;
unsigned int periods;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
/* Attempt to match 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);
}
/* Need to at least double buffer */
periods = 2;
status = ALSA_snd_pcm_hw_params_set_periods_min(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
}
status = ALSA_snd_pcm_hw_params_set_periods_first(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
}
/* "set" the hardware with the desired parameters */
status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
@ -446,25 +500,13 @@ ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
return(-1);
}
/* 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->spec.samples = persize;
/* This is useful for debugging */
if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {
snd_pcm_uframes_t persize = 0;
unsigned int periods = 0;
snd_pcm_uframes_t bufsize;
ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL);
ALSA_snd_pcm_hw_params_get_periods(hwparams, &periods, NULL);
ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
fprintf(stderr,
"ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
@ -474,78 +516,6 @@ ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
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)
{
@ -692,14 +662,11 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
this->spec.freq = rate;
/* Set the buffer size, in samples */
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));
}
status = ALSA_set_buffer_size(this, hwparams);
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);
@ -737,9 +704,11 @@ 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;
@ -828,7 +797,7 @@ ALSA_HotplugThread(void *arg)
ALSA_Device *seen;
ALSA_Device *prev;
if (ALSA_snd_device_name_hint(-1, "pcm", &hints) != -1) {
if (ALSA_snd_device_name_hint(-1, "pcm", &hints) == 0) {
int i, j;
const char *match = NULL;
int bestmatch = 0xFFFF;

2
Engine/lib/sdl/src/audio/alsa/SDL_alsa_audio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

27
Engine/lib/sdl/src/audio/android/SDL_androidaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -57,7 +57,9 @@ 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_S16LSB)) {
if ((test_format == AUDIO_U8) ||
(test_format == AUDIO_S16) ||
(test_format == AUDIO_F32)) {
this->spec.format = test_format;
break;
}
@ -69,25 +71,8 @@ ANDROIDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
return SDL_SetError("No compatible audio format!");
}
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!");
if (Android_JNI_OpenAudioDevice(iscapture, &this->spec) < 0) {
return -1;
}
SDL_CalculateAudioSpec(&this->spec);

2
Engine/lib/sdl/src/audio/android/SDL_androidaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

4
Engine/lib/sdl/src/audio/arts/SDL_artsaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -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

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

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/coreaudio/SDL_coreaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

13
Engine/lib/sdl/src/audio/coreaudio/SDL_coreaudio.m
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -354,7 +354,7 @@ static BOOL update_audio_session(_THIS, SDL_bool open)
return NO;
}
if (open_playback_devices + open_capture_devices == 1) {
if (open && (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);
@ -376,25 +376,26 @@ static BOOL update_audio_session(_THIS, SDL_bool open)
/* An interruption end notification is not guaranteed to be sent if
we were previously interrupted... resuming if needed when the app
becomes active seems to be the way to go. */
// Note: object: below needs to be nil, as otherwise it filters by the object, and session doesn't send foreground / active notifications. johna
[center addObserver:listener
selector:@selector(applicationBecameActive:)
name:UIApplicationDidBecomeActiveNotification
object:session];
object:nil];
[center addObserver:listener
selector:@selector(applicationBecameActive:)
name:UIApplicationWillEnterForegroundNotification
object:session];
object:nil];
this->hidden->interruption_listener = CFBridgingRetain(listener);
} else {
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];
}
}
}
@ -728,6 +729,8 @@ audioqueue_thread(void *arg)
return 0;
}
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
/* init was successful, alert parent thread and start running... */
SDL_SemPost(this->hidden->ready_semaphore);
while (!SDL_AtomicGet(&this->hidden->shutdown)) {

8
Engine/lib/sdl/src/audio/directsound/SDL_directsound.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -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;

2
Engine/lib/sdl/src/audio/directsound/SDL_directsound.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/disk/SDL_diskaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/disk/SDL_diskaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/dsp/SDL_dspaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/dsp/SDL_dspaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/dummy/SDL_dummyaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/dummy/SDL_dummyaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

26
Engine/lib/sdl/src/audio/emscripten/SDL_emscriptenaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -35,6 +35,7 @@ FeedAudioDevice(_THIS, const void *buf, const int buflen)
{
const int framelen = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
EM_ASM_ARGS({
var SDL2 = Module['SDL2'];
var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
@ -100,6 +101,7 @@ HandleCaptureProcess(_THIS)
}
EM_ASM_ARGS({
var SDL2 = Module['SDL2'];
var numChannels = SDL2.capture.currentCaptureBuffer.numberOfChannels;
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
@ -145,6 +147,7 @@ static void
EMSCRIPTENAUDIO_CloseDevice(_THIS)
{
EM_ASM_({
var SDL2 = Module['SDL2'];
if ($0) {
if (SDL2.capture.silenceTimer !== undefined) {
clearTimeout(SDL2.capture.silenceTimer);
@ -196,11 +199,12 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
/* based on parts of library_sdl.js */
/* create context (TODO: this puts stuff in the global namespace...)*/
/* create context */
result = EM_ASM_INT({
if(typeof(SDL2) === 'undefined') {
SDL2 = {};
if(typeof(Module['SDL2']) === 'undefined') {
Module['SDL2'] = {};
}
var SDL2 = Module['SDL2'];
if (!$0) {
SDL2.audio = {};
} else {
@ -246,9 +250,13 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
}
SDL_zerop(this->hidden);
#endif
this->hidden = (struct SDL_PrivateAudioData *)0x1;
/* limit to native freq */
this->spec.freq = EM_ASM_INT_V({ return SDL2.audioContext.sampleRate; });
this->spec.freq = EM_ASM_INT_V({
var SDL2 = Module['SDL2'];
return SDL2.audioContext.sampleRate;
});
SDL_CalculateAudioSpec(&this->spec);
@ -270,6 +278,7 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
to be honest. */
EM_ASM_({
var SDL2 = Module['SDL2'];
var have_microphone = function(stream) {
//console.log('SDL audio capture: we have a microphone! Replacing silence callback.');
if (SDL2.capture.silenceTimer !== undefined) {
@ -282,7 +291,7 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
if ((SDL2 === undefined) || (SDL2.capture === undefined)) { return; }
audioProcessingEvent.outputBuffer.getChannelData(0).fill(0.0);
SDL2.capture.currentCaptureBuffer = audioProcessingEvent.inputBuffer;
Runtime.dynCall('vi', $2, [$3]);
dynCall('vi', $2, [$3]);
};
SDL2.capture.mediaStreamNode.connect(SDL2.capture.scriptProcessorNode);
SDL2.capture.scriptProcessorNode.connect(SDL2.audioContext.destination);
@ -298,7 +307,7 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
SDL2.capture.silenceBuffer.getChannelData(0).fill(0.0);
var silence_callback = function() {
SDL2.capture.currentCaptureBuffer = SDL2.capture.silenceBuffer;
Runtime.dynCall('vi', $2, [$3]);
dynCall('vi', $2, [$3]);
};
SDL2.capture.silenceTimer = setTimeout(silence_callback, ($1 / SDL2.audioContext.sampleRate) * 1000);
@ -312,11 +321,12 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
} else {
/* setup a ScriptProcessorNode */
EM_ASM_ARGS({
var SDL2 = Module['SDL2'];
SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
if ((SDL2 === undefined) || (SDL2.audio === undefined)) { return; }
SDL2.audio.currentOutputBuffer = e['outputBuffer'];
Runtime.dynCall('vi', $2, [$3]);
dynCall('vi', $2, [$3]);
};
SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);

2
Engine/lib/sdl/src/audio/emscripten/SDL_emscriptenaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/esd/SDL_esdaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/esd/SDL_esdaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/fusionsound/SDL_fsaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/fusionsound/SDL_fsaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/haiku/SDL_haikuaudio.cc
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/haiku/SDL_haikuaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

37
Engine/lib/sdl/src/audio/jack/SDL_jackaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -44,7 +44,9 @@ 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 *);
@ -135,7 +137,9 @@ 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;
@ -273,10 +277,6 @@ JACK_CloseDevice(_THIS)
SDL_DestroySemaphore(this->hidden->iosem);
}
if (this->hidden->devports) {
JACK_jack_free(this->hidden->devports);
}
SDL_free(this->hidden->iobuffer);
}
@ -292,9 +292,11 @@ 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 */
@ -311,15 +313,30 @@ 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[++channels]) {
while (devports[++ports]) {
/* 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. */
@ -368,16 +385,16 @@ 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[i] : sdlport;
const char *dstport = iscapture ? sdlport : devports[i];
const char *srcport = iscapture ? devports[audio_ports[i]] : sdlport;
const char *dstport = iscapture ? sdlport : devports[audio_ports[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;

3
Engine/lib/sdl/src/audio/jack/SDL_jackaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -33,7 +33,6 @@ 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/nacl/SDL_naclaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/nacl/SDL_naclaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/nas/SDL_nasaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/nas/SDL_nasaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

171
Engine/lib/sdl/src/audio/netbsd/SDL_netbsdaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -43,12 +43,7 @@
#include "../SDL_audiodev_c.h"
#include "SDL_netbsdaudio.h"
/* Use timer for synchronization */
/* #define USE_TIMER_SYNC */
/* #define DEBUG_AUDIO */
/* #define DEBUG_AUDIO_STREAM */
static void
NETBSDAUDIO_DetectDevices(void)
@ -63,14 +58,14 @@ NETBSDAUDIO_Status(_THIS)
#ifdef DEBUG_AUDIO
/* *INDENT-OFF* */
audio_info_t info;
const audio_prinfo *prinfo;
const struct audio_prinfo *prinfo;
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
fprintf(stderr, "AUDIO_GETINFO failed.\n");
return;
}
prinfo = this->iscapture ? &info.play : &info.record;
prinfo = this->iscapture ? &info.record : &info.play;
fprintf(stderr, "\n"
"[%s info]\n"
@ -115,90 +110,37 @@ NETBSDAUDIO_Status(_THIS)
(info.mode == AUMODE_PLAY) ? "PLAY"
: (info.mode = AUMODE_RECORD) ? "RECORD"
: (info.mode == AUMODE_PLAY_ALL ? "PLAY_ALL" : "?"));
fprintf(stderr, "\n"
"[audio spec]\n"
"format : 0x%x\n"
"size : %u\n"
"",
this->spec.format,
this->spec.size);
/* *INDENT-ON* */
#endif /* DEBUG_AUDIO */
}
/* This function waits until it is possible to write a full sound buffer */
static void
NETBSDAUDIO_WaitDevice(_THIS)
{
#ifndef USE_BLOCKING_WRITES /* Not necessary when using blocking writes */
/* See if we need to use timed audio synchronization */
if (this->hidden->frame_ticks) {
/* Use timer for general audio synchronization */
Sint32 ticks;
ticks = ((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
}
} else {
/* Use SDL_IOReady() for audio synchronization */
#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
if (SDL_IOReady(this->hidden->audio_fd, SDL_TRUE, 10 * 1000)
<= 0) {
const char *message =
"Audio timeout - buggy audio driver? (disabled)";
/* In general we should never print to the screen,
but in this case we have no other way of letting
the user know what happened.
*/
fprintf(stderr, "SDL: %s\n", message);
SDL_OpenedAudioDeviceDisconnected(this);
/* Don't try to close - may hang */
this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Done disabling audio\n");
#endif
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Ready!\n");
#endif
}
#endif /* !USE_BLOCKING_WRITES */
}
static void
NETBSDAUDIO_PlayDevice(_THIS)
{
int written, p = 0;
struct SDL_PrivateAudioData *h = this->hidden;
int written;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
do {
written = write(this->hidden->audio_fd,
&this->hidden->mixbuf[p], this->hidden->mixlen - p);
if (written > 0)
p += written;
if (written == -1 && errno != 0 && errno != EAGAIN && errno != EINTR) {
/* Non recoverable error has occurred. It should be reported!!! */
perror("audio");
break;
}
/* Write the audio data */
written = write(h->audio_fd, h->mixbuf, h->mixlen);
if (written == -1) {
/* Non recoverable error has occurred. It should be reported!!! */
SDL_OpenedAudioDeviceDisconnected(this);
perror("audio");
return;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
if (p < this->hidden->mixlen
|| ((written < 0) && ((errno == 0) || (errno == EAGAIN)))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while (p < this->hidden->mixlen);
/* If timer synchronization is enabled, set the next write frame */
if (this->hidden->frame_ticks) {
this->hidden->next_frame += this->hidden->frame_ticks;
}
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
SDL_OpenedAudioDeviceDisconnected(this);
}
}
static Uint8 *
@ -212,28 +154,19 @@ static int
NETBSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
{
Uint8 *buffer = (Uint8 *) _buffer;
int br, p = 0;
int br;
/* Capture the audio data, checking for EAGAIN on broken audio drivers */
do {
br = read(this->hidden->audio_fd, buffer + p, buflen - p);
if (br > 0)
p += br;
if (br == -1 && errno != 0 && errno != EAGAIN && errno != EINTR) {
/* Non recoverable error has occurred. It should be reported!!! */
perror("audio");
return p ? p : -1;
}
br = read(this->hidden->audio_fd, buffer, buflen);
if (br == -1) {
/* Non recoverable error has occurred. It should be reported!!! */
perror("audio");
return -1;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Captured %d bytes of audio data\n", br);
fprintf(stderr, "Captured %d bytes of audio data\n", br);
#endif
if (p < buflen
|| ((br < 0) && ((errno == 0) || (errno == EAGAIN)))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while (p < buflen);
return 0;
}
static void
@ -271,10 +204,9 @@ NETBSDAUDIO_CloseDevice(_THIS)
static int
NETBSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
SDL_AudioFormat format = 0;
audio_info_t info;
audio_prinfo *prinfo = iscapture ? &info.play : &info.record;
struct audio_prinfo *prinfo = iscapture ? &info.record : &info.play;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
@ -294,25 +226,16 @@ NETBSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_zerop(this->hidden);
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
this->hidden->audio_fd = open(devname, iscapture ? O_RDONLY : O_WRONLY);
if (this->hidden->audio_fd < 0) {
return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
}
AUDIO_INITINFO(&info);
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
prinfo->encoding = AUDIO_ENCODING_NONE;
/* Set to play mode */
info.mode = iscapture ? AUMODE_RECORD : AUMODE_PLAY;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) < 0) {
return SDL_SetError("Couldn't put device into play mode");
}
AUDIO_INITINFO(&info);
for (format = SDL_FirstAudioFormat(this->spec.format);
format; format = SDL_NextAudioFormat()) {
for (format = SDL_FirstAudioFormat(this->spec.format); format;) {
switch (format) {
case AUDIO_U8:
prinfo->encoding = AUDIO_ENCODING_ULINEAR;
@ -338,34 +261,33 @@ NETBSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
prinfo->encoding = AUDIO_ENCODING_ULINEAR_BE;
prinfo->precision = 16;
break;
default:
continue;
}
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {
if (prinfo->encoding != AUDIO_ENCODING_NONE) {
break;
}
format = SDL_NextAudioFormat();
}
if (!format) {
if (prinfo->encoding == AUDIO_ENCODING_NONE) {
return SDL_SetError("No supported encoding for 0x%x", this->spec.format);
}
this->spec.format = format;
AUDIO_INITINFO(&info);
prinfo->channels = this->spec.channels;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == -1) {
this->spec.channels = 1;
}
AUDIO_INITINFO(&info);
prinfo->sample_rate = this->spec.freq;
/* Calculate spec parameters based on our chosen format */
SDL_CalculateAudioSpec(&this->spec);
info.mode = iscapture ? AUMODE_RECORD : AUMODE_PLAY;
info.blocksize = this->spec.size;
info.hiwat = 5;
info.lowat = 3;
prinfo->sample_rate = this->spec.freq;
prinfo->channels = this->spec.channels;
(void) ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info);
(void) ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info);
this->spec.freq = prinfo->sample_rate;
this->spec.channels = prinfo->channels;
if (!iscapture) {
/* Allocate mixing buffer */
@ -390,7 +312,6 @@ NETBSDAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->DetectDevices = NETBSDAUDIO_DetectDevices;
impl->OpenDevice = NETBSDAUDIO_OpenDevice;
impl->PlayDevice = NETBSDAUDIO_PlayDevice;
impl->WaitDevice = NETBSDAUDIO_WaitDevice;
impl->GetDeviceBuf = NETBSDAUDIO_GetDeviceBuf;
impl->CloseDevice = NETBSDAUDIO_CloseDevice;
impl->CaptureFromDevice = NETBSDAUDIO_CaptureFromDevice;

2
Engine/lib/sdl/src/audio/netbsd/SDL_netbsdaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

643
Engine/lib/sdl/src/audio/openslES/SDL_openslES.c Normal file
View file

@ -0,0 +1,643 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2019 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_OPENSLES
/* For more discussion of low latency audio on Android, see this:
https://googlesamples.github.io/android-audio-high-performance/guides/opensl_es.html
*/
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_openslES.h"
/* for native audio */
#include <SLES/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>
#include <android/log.h>
#define LOG_TAG "SDL_openslES"
#if 0
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR,LOG_TAG,__VA_ARGS__)
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__)
//#define LOGV(...) __android_log_print(ANDROID_LOG_VERBOSE,LOG_TAG,__VA_ARGS__)
#define LOGV(...)
#else
#define LOGE(...)
#define LOGI(...)
#define LOGV(...)
#endif
/* engine interfaces */
static SLObjectItf engineObject = NULL;
static SLEngineItf engineEngine = NULL;
/* output mix interfaces */
static SLObjectItf outputMixObject = NULL;
// static SLEnvironmentalReverbItf outputMixEnvironmentalReverb = NULL;
/* aux effect on the output mix, used by the buffer queue player */
/* static const SLEnvironmentalReverbSettings reverbSettings = SL_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR; */
/* buffer queue player interfaces */
static SLObjectItf bqPlayerObject = NULL;
static SLPlayItf bqPlayerPlay = NULL;
static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue = NULL;
#if 0
static SLEffectSendItf bqPlayerEffectSend = NULL;
static SLMuteSoloItf bqPlayerMuteSolo = NULL;
static SLVolumeItf bqPlayerVolume = NULL;
#endif
#if 0
/* recorder interfaces TODO */
static SLObjectItf recorderObject = NULL;
static SLRecordItf recorderRecord;
static SLAndroidSimpleBufferQueueItf recorderBufferQueue;
#endif
/* pointer and size of the next player buffer to enqueue, and number of remaining buffers */
#if 0
static short *nextBuffer;
static unsigned nextSize;
static int nextCount;
#endif
// static SDL_AudioDevice* audioDevice = NULL;
#if 0
static const char *sldevaudiorecorderstr = "SLES Audio Recorder";
static const char *sldevaudioplayerstr = "SLES Audio Player";
#define SLES_DEV_AUDIO_RECORDER sldevaudiorecorderstr
#define SLES_DEV_AUDIO_PLAYER sldevaudioplayerstr
static void openslES_DetectDevices( int iscapture )
{
LOGI( "openSLES_DetectDevices()" );
if ( iscapture )
addfn( SLES_DEV_AUDIO_RECORDER );
else
addfn( SLES_DEV_AUDIO_PLAYER );
return;
}
#endif
static void openslES_DestroyEngine();
static int
openslES_CreateEngine()
{
SLresult result;
LOGI("openSLES_CreateEngine()");
/* create engine */
result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
if (SL_RESULT_SUCCESS != result) {
LOGE("slCreateEngine failed");
goto error;
}
LOGI("slCreateEngine OK");
/* realize the engine */
result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
if (SL_RESULT_SUCCESS != result) {
LOGE("RealizeEngine failed");
goto error;
}
LOGI("RealizeEngine OK");
/* get the engine interface, which is needed in order to create other objects */
result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
if (SL_RESULT_SUCCESS != result) {
LOGE("EngineGetInterface failed");
goto error;
}
LOGI("EngineGetInterface OK");
/* create output mix, with environmental reverb specified as a non-required interface */
/* const SLInterfaceID ids[1] = { SL_IID_ENVIRONMENTALREVERB }; */
/* const SLboolean req[1] = { SL_BOOLEAN_FALSE }; */
const SLInterfaceID ids[1] = { SL_IID_VOLUME };
const SLboolean req[1] = { SL_BOOLEAN_FALSE };
result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, ids, req);
if (SL_RESULT_SUCCESS != result) {
LOGE("CreateOutputMix failed");
goto error;
}
LOGI("CreateOutputMix OK");
/* realize the output mix */
result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
if (SL_RESULT_SUCCESS != result) {
LOGE("RealizeOutputMix failed");
goto error;
}
return 1;
error:
openslES_DestroyEngine();
return 0;
}
static void openslES_DestroyPCMPlayer(_THIS);
static void openslES_DestroyPCMRecorder(_THIS);
static void openslES_DestroyEngine()
{
LOGI("openslES_DestroyEngine()");
// openslES_DestroyPCMPlayer(this);
// openslES_DestroyPCMRecorder(this);
/* destroy output mix object, and invalidate all associated interfaces */
if (outputMixObject != NULL) {
(*outputMixObject)->Destroy(outputMixObject);
outputMixObject = NULL;
/* outputMixEnvironmentalReverb = NULL; */
}
/* destroy engine object, and invalidate all associated interfaces */
if (engineObject != NULL) {
(*engineObject)->Destroy(engineObject);
engineObject = NULL;
engineEngine = NULL;
}
return;
}
/* this callback handler is called every time a buffer finishes playing */
static void
bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) context;
LOGV("SLES: Playback Callmeback");
SDL_SemPost(audiodata->playsem);
return;
}
static int
openslES_CreatePCMRecorder(_THIS)
{
/* struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden; */
LOGE("openslES_CreatePCMRecorder not implimented yet!");
return SDL_SetError("openslES_CreatePCMRecorder not implimented yet!");
}
static void
openslES_DestroyPCMRecorder(_THIS)
{
/* struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden; */
return;
}
static int
openslES_CreatePCMPlayer(_THIS)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden;
SLDataFormat_PCM format_pcm;
SLresult result;
int i;
/* If we want to add floating point audio support (requires API level 21)
it can be done as described here:
https://developer.android.com/ndk/guides/audio/opensl/android-extensions.html#floating-point
*/
#if 1
/* Just go with signed 16-bit audio as it's the most compatible */
this->spec.format = AUDIO_S16SYS;
#else
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) {
if (SDL_AUDIO_ISSIGNED(test_format) && SDL_AUDIO_ISINT(test_format)) {
break;
}
test_format = SDL_NextAudioFormat();
}
if (test_format == 0) {
/* Didn't find a compatible format : */
LOGI( "No compatible audio format, using signed 16-bit audio" );
test_format = AUDIO_S16SYS;
}
this->spec.format = test_format;
#endif
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
LOGI("Try to open %u hz %u bit chan %u %s samples %u",
this->spec.freq, SDL_AUDIO_BITSIZE(this->spec.format),
this->spec.channels, (this->spec.format & 0x1000) ? "BE" : "LE", this->spec.samples);
/* configure audio source */
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, NUM_BUFFERS };
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = this->spec.channels;
format_pcm.samplesPerSec = this->spec.freq * 1000; /* / kilo Hz to milli Hz */
format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
format_pcm.containerSize = SDL_AUDIO_BITSIZE(this->spec.format);
if (SDL_AUDIO_ISBIGENDIAN(this->spec.format)) {
format_pcm.endianness = SL_BYTEORDER_BIGENDIAN;
} else {
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
}
/*
#define SL_SPEAKER_FRONT_LEFT ((SLuint32) 0x00000001)
#define SL_SPEAKER_FRONT_RIGHT ((SLuint32) 0x00000002)
#define SL_SPEAKER_FRONT_CENTER ((SLuint32) 0x00000004)
#define SL_SPEAKER_LOW_FREQUENCY ((SLuint32) 0x00000008)
#define SL_SPEAKER_BACK_LEFT ((SLuint32) 0x00000010)
#define SL_SPEAKER_BACK_RIGHT ((SLuint32) 0x00000020)
#define SL_SPEAKER_FRONT_LEFT_OF_CENTER ((SLuint32) 0x00000040)
#define SL_SPEAKER_FRONT_RIGHT_OF_CENTER ((SLuint32) 0x00000080)
#define SL_SPEAKER_BACK_CENTER ((SLuint32) 0x00000100)
#define SL_SPEAKER_SIDE_LEFT ((SLuint32) 0x00000200)
#define SL_SPEAKER_SIDE_RIGHT ((SLuint32) 0x00000400)
#define SL_SPEAKER_TOP_CENTER ((SLuint32) 0x00000800)
#define SL_SPEAKER_TOP_FRONT_LEFT ((SLuint32) 0x00001000)
#define SL_SPEAKER_TOP_FRONT_CENTER ((SLuint32) 0x00002000)
#define SL_SPEAKER_TOP_FRONT_RIGHT ((SLuint32) 0x00004000)
#define SL_SPEAKER_TOP_BACK_LEFT ((SLuint32) 0x00008000)
#define SL_SPEAKER_TOP_BACK_CENTER ((SLuint32) 0x00010000)
#define SL_SPEAKER_TOP_BACK_RIGHT ((SLuint32) 0x00020000)
*/
#define SL_ANDROID_SPEAKER_STEREO (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT)
#define SL_ANDROID_SPEAKER_QUAD (SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT)
#define SL_ANDROID_SPEAKER_5DOT1 (SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY)
#define SL_ANDROID_SPEAKER_7DOT1 (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT | SL_SPEAKER_SIDE_RIGHT)
switch (this->spec.channels)
{
case 1:
format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT;
break;
case 2:
format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO;
break;
case 3:
format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_FRONT_CENTER;
break;
case 4:
format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD;
break;
case 5:
format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER;
break;
case 6:
format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1;
break;
case 7:
format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_BACK_CENTER;
break;
case 8:
format_pcm.channelMask = SL_ANDROID_SPEAKER_7DOT1;
break;
default:
/* Unknown number of channels, fall back to stereo */
this->spec.channels = 2;
format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
break;
}
SLDataSource audioSrc = { &loc_bufq, &format_pcm };
/* configure audio sink */
SLDataLocator_OutputMix loc_outmix = { SL_DATALOCATOR_OUTPUTMIX, outputMixObject };
SLDataSink audioSnk = { &loc_outmix, NULL };
/* create audio player */
const SLInterfaceID ids[2] = {
SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
SL_IID_VOLUME
};
const SLboolean req[2] = {
SL_BOOLEAN_TRUE,
SL_BOOLEAN_FALSE,
};
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
if (SL_RESULT_SUCCESS != result) {
LOGE("CreateAudioPlayer failed");
goto failed;
}
/* realize the player */
result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
if (SL_RESULT_SUCCESS != result) {
LOGE("RealizeAudioPlayer failed");
goto failed;
}
/* get the play interface */
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
if (SL_RESULT_SUCCESS != result) {
LOGE("SL_IID_PLAY interface get failed");
goto failed;
}
/* get the buffer queue interface */
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bqPlayerBufferQueue);
if (SL_RESULT_SUCCESS != result) {
LOGE("SL_IID_BUFFERQUEUE interface get failed");
goto failed;
}
/* register callback on the buffer queue */
/* context is '(SDL_PrivateAudioData *)this->hidden' */
result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, this->hidden);
if (SL_RESULT_SUCCESS != result) {
LOGE("RegisterCallback failed");
goto failed;
}
#if 0
/* get the effect send interface */
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND, &bqPlayerEffectSend);
if (SL_RESULT_SUCCESS != result)
{
LOGE("SL_IID_EFFECTSEND interface get failed");
goto failed;
}
#endif
#if 0 /* mute/solo is not supported for sources that are known to be mono, as this is */
/* get the mute/solo interface */
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_MUTESOLO, &bqPlayerMuteSolo);
assert(SL_RESULT_SUCCESS == result);
(void) result;
#endif
#if 0
/* get the volume interface */
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
if (SL_RESULT_SUCCESS != result) {
LOGE("SL_IID_VOLUME interface get failed");
/* goto failed; */
}
#endif
/* Create the audio buffer semaphore */
audiodata->playsem = SDL_CreateSemaphore(NUM_BUFFERS - 1);
if (!audiodata->playsem) {
LOGE("cannot create Semaphore!");
goto failed;
}
/* Create the sound buffers */
audiodata->mixbuff = (Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
if (audiodata->mixbuff == NULL) {
LOGE("mixbuffer allocate - out of memory");
goto failed;
}
for (i = 0; i < NUM_BUFFERS; i++) {
audiodata->pmixbuff[i] = audiodata->mixbuff + i * this->spec.size;
}
/* set the player's state to playing */
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
if (SL_RESULT_SUCCESS != result) {
LOGE("Play set state failed");
goto failed;
}
return 0;
failed:
openslES_DestroyPCMPlayer(this);
return SDL_SetError("Open device failed!");
}
static void
openslES_DestroyPCMPlayer(_THIS)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden;
SLresult result;
/* set the player's state to 'stopped' */
if (bqPlayerPlay != NULL) {
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_STOPPED);
if (SL_RESULT_SUCCESS != result) {
SDL_SetError("Stopped set state failed");
}
}
/* destroy buffer queue audio player object, and invalidate all associated interfaces */
if (bqPlayerObject != NULL) {
(*bqPlayerObject)->Destroy(bqPlayerObject);
bqPlayerObject = NULL;
bqPlayerPlay = NULL;
bqPlayerBufferQueue = NULL;
#if 0
bqPlayerEffectSend = NULL;
bqPlayerMuteSolo = NULL;
bqPlayerVolume = NULL;
#endif
}
if (audiodata->playsem) {
SDL_DestroySemaphore(audiodata->playsem);
audiodata->playsem = NULL;
}
if (audiodata->mixbuff) {
SDL_free(audiodata->mixbuff);
}
return;
}
static int
openslES_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
if (iscapture) {
LOGI("openslES_OpenDevice( ) %s for capture", devname);
return openslES_CreatePCMRecorder(this);
} else {
LOGI("openslES_OpenDevice( ) %s for playing", devname);
return openslES_CreatePCMPlayer(this);
}
}
static void
openslES_CloseDevice(_THIS)
{
/* struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden; */
if (this->iscapture) {
LOGI("openslES_CloseDevice( ) for capture");
openslES_DestroyPCMRecorder(this);
} else {
LOGI("openslES_CloseDevice( ) for playing");
openslES_DestroyPCMPlayer(this);
}
SDL_free(this->hidden);
return;
}
static void
openslES_WaitDevice(_THIS)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden;
LOGV("openslES_WaitDevice( )");
/* Wait for an audio chunk to finish */
/* WaitForSingleObject(this->hidden->audio_sem, INFINITE); */
SDL_SemWait(audiodata->playsem);
return;
}
/*/ n playn sem */
/* getbuf 0 - 1 */
/* fill buff 0 - 1 */
/* play 0 - 0 1 */
/* wait 1 0 0 */
/* getbuf 1 0 0 */
/* fill buff 1 0 0 */
/* play 0 0 0 */
/* wait */
/* */
/* okay.. */
static Uint8 *
openslES_GetDeviceBuf(_THIS)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden;
LOGV("openslES_GetDeviceBuf( )");
return audiodata->pmixbuff[audiodata->next_buffer];
}
static void
openslES_PlayDevice(_THIS)
{
struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *) this->hidden;
SLresult result;
LOGV("======openslES_PlayDevice( )======");
/* Queue it up */
result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, audiodata->pmixbuff[audiodata->next_buffer], this->spec.size);
audiodata->next_buffer++;
if (audiodata->next_buffer >= NUM_BUFFERS) {
audiodata->next_buffer = 0;
}
/* If Enqueue fails, callback won't be called.
* Post the semphore, not to run out of buffer */
if (SL_RESULT_SUCCESS != result) {
SDL_SemPost(audiodata->playsem);
}
return;
}
static int
openslES_Init(SDL_AudioDriverImpl * impl)
{
LOGI("openslES_Init() called");
if (!openslES_CreateEngine()) {
return 0;
}
LOGI("openslES_Init() - set pointers");
/* Set the function pointers */
/* impl->DetectDevices = openslES_DetectDevices; */
impl->OpenDevice = openslES_OpenDevice;
impl->CloseDevice = openslES_CloseDevice;
impl->PlayDevice = openslES_PlayDevice;
impl->GetDeviceBuf = openslES_GetDeviceBuf;
impl->Deinitialize = openslES_DestroyEngine;
impl->WaitDevice = openslES_WaitDevice;
/* and the capabilities */
impl->HasCaptureSupport = 0; /* TODO */
impl->OnlyHasDefaultOutputDevice = 1;
/* impl->OnlyHasDefaultInputDevice = 1; */
LOGI("openslES_Init() - succes");
/* this audio target is available. */
return 1;
}
AudioBootStrap openslES_bootstrap = {
"openslES", "opensl ES audio driver", openslES_Init, 0
};
void openslES_ResumeDevices()
{
if (bqPlayerPlay != NULL) {
/* set the player's state to 'playing' */
SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
if (SL_RESULT_SUCCESS != result) {
SDL_SetError("openslES_ResumeDevices failed");
}
}
}
void openslES_PauseDevices()
{
if (bqPlayerPlay != NULL) {
/* set the player's state to 'paused' */
SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PAUSED);
if (SL_RESULT_SUCCESS != result) {
SDL_SetError("openslES_PauseDevices failed");
}
}
}
#endif /* SDL_AUDIO_DRIVER_OPENSLES */
/* vi: set ts=4 sw=4 expandtab: */

50
Engine/lib/sdl/src/audio/openslES/SDL_openslES.h Normal file
View file

@ -0,0 +1,50 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2019 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_openslesaudio_h
#define _SDL_openslesaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2 /* -- Don't lower this! */
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
Uint8 *mixbuff;
int next_buffer;
Uint8 *pmixbuff[NUM_BUFFERS];
SDL_sem *playsem;
#if 0
SDL_sem *recsem;
#endif
};
void openslES_ResumeDevices(void);
void openslES_PauseDevices(void);
#endif /* _SDL_openslesaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

2
Engine/lib/sdl/src/audio/paudio/SDL_paudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/paudio/SDL_paudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/psp/SDL_pspaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/psp/SDL_pspaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

4
Engine/lib/sdl/src/audio/pulseaudio/SDL_pulseaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -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 *);

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

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/qsa/SDL_qsa_audio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/qsa/SDL_qsa_audio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/sndio/SDL_sndioaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/sndio/SDL_sndioaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/sun/SDL_sunaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

2
Engine/lib/sdl/src/audio/sun/SDL_sunaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

30
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -161,21 +161,6 @@ WASAPI_DetectDevices(void)
WASAPI_EnumerateEndpoints();
}
static int
WASAPI_GetPendingBytes(_THIS)
{
UINT32 frames = 0;
/* it's okay to fail here; we'll deal with failures in the audio thread. */
/* FIXME: need a lock around checking this->hidden->client */
if (this->hidden->client != NULL) { /* definitely activated? */
if (FAILED(IAudioClient_GetCurrentPadding(this->hidden->client, &frames))) {
return 0; /* oh well. */
}
}
return ((int) frames) * this->hidden->framesize;
}
static SDL_INLINE SDL_bool
WasapiFailed(_THIS, const HRESULT err)
{
@ -327,8 +312,8 @@ static void
WASAPI_WaitDevice(_THIS)
{
while (RecoverWasapiIfLost(this) && this->hidden->client && this->hidden->event) {
/*SDL_Log("WAITDEVICE");*/
if (WaitForSingleObjectEx(this->hidden->event, INFINITE, FALSE) == WAIT_OBJECT_0) {
DWORD waitResult = WaitForSingleObjectEx(this->hidden->event, 200, FALSE);
if (waitResult == WAIT_OBJECT_0) {
const UINT32 maxpadding = this->spec.samples;
UINT32 padding = 0;
if (!WasapiFailed(this, IAudioClient_GetCurrentPadding(this->hidden->client, &padding))) {
@ -337,7 +322,7 @@ WASAPI_WaitDevice(_THIS)
break;
}
}
} else {
} else if (waitResult != WAIT_TIMEOUT) {
/*SDL_Log("WASAPI FAILED EVENT!");*/
IAudioClient_Stop(this->hidden->client);
SDL_OpenedAudioDeviceDisconnected(this);
@ -725,6 +710,12 @@ WASAPI_ThreadDeinit(_THIS)
WASAPI_PlatformThreadDeinit(this);
}
void
WASAPI_BeginLoopIteration(_THIS)
{
/* no-op. */
}
static void
WASAPI_Deinitialize(void)
{
@ -759,7 +750,6 @@ WASAPI_Init(SDL_AudioDriverImpl * impl)
impl->OpenDevice = WASAPI_OpenDevice;
impl->PlayDevice = WASAPI_PlayDevice;
impl->WaitDevice = WASAPI_WaitDevice;
impl->GetPendingBytes = WASAPI_GetPendingBytes;
impl->GetDeviceBuf = WASAPI_GetDeviceBuf;
impl->CaptureFromDevice = WASAPI_CaptureFromDevice;
impl->FlushCapture = WASAPI_FlushCapture;

2
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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

70
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi_win32.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -351,10 +351,42 @@ 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"
@ -369,22 +401,36 @@ 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))) {
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);
if (SUCCEEDED(IMMDevice_GetId(device, &item->devid))) {
item->devname = GetWasapiDeviceName(device);
}
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);
}
@ -405,12 +451,6 @@ 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: */

67
Engine/lib/sdl/src/audio/wasapi/SDL_wasapi_winrt.cpp
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -185,20 +185,9 @@ struct SDL_WasapiActivationHandler : public RuntimeClass< RuntimeClassFlags< Cla
HRESULT
SDL_WasapiActivationHandler::ActivateCompleted(IActivateAudioInterfaceAsyncOperation *async)
{
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);
}
}
// 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;
}
@ -236,27 +225,47 @@ WASAPI_ActivateDevice(_THIS, const SDL_bool isrecovery)
IActivateAudioInterfaceAsyncOperation *async = nullptr;
const HRESULT ret = ActivateAudioInterfaceAsync(devid, __uuidof(IAudioClient), nullptr, handler.Get(), &async);
if (async != nullptr) {
async->Release();
}
if (FAILED(ret)) {
if (FAILED(ret) || async == nullptr) {
if (async != nullptr) {
async->Release();
}
handler.Get()->Release();
WASAPI_UnrefDevice(_this);
return WIN_SetErrorFromHRESULT("WASAPI can't activate requested audio endpoint", ret);
}
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);
}
/* 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

7
Engine/lib/sdl/src/audio/winmm/SDL_winmm.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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
@ -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)); \
@ -375,8 +375,7 @@ 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");
}

2
Engine/lib/sdl/src/audio/winmm/SDL_winmm.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2019 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