Hash :
b48e54aa
Author :
Date :
2015-01-26T22:00:29
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652 653 654 655 656 657 658 659 660 661 662 663 664 665 666 667 668 669 670 671 672 673 674 675 676 677 678 679 680 681 682 683 684 685
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_ALSA
/* Allow access to a raw mixing buffer */
#include <sys/types.h>
#include <signal.h> /* For kill() */
#include <errno.h>
#include <string.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_alsa_audio.h"
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
#include "SDL_loadso.h"
#endif
static int (*ALSA_snd_pcm_open)
(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);
static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
(snd_pcm_t *, const void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);
static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);
static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);
static const char *(*ALSA_snd_strerror) (int);
static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void);
static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void);
static void (*ALSA_snd_pcm_hw_params_copy)
(snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_set_access)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
static int (*ALSA_snd_pcm_hw_params_set_format)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
static int (*ALSA_snd_pcm_hw_params_set_channels)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
static int (*ALSA_snd_pcm_hw_params_get_channels)
(const snd_pcm_hw_params_t *, unsigned int *);
static int (*ALSA_snd_pcm_hw_params_set_rate_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
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)
(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 *);
static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)
(snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params_get_buffer_size)
(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *,
snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_sw_params_set_start_threshold)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);
static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof
#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof
static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
static void *alsa_handle = NULL;
static int
load_alsa_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(alsa_handle, fn);
if (*addr == NULL) {
/* Don't call SDL_SetError(): SDL_LoadFunction already did. */
return 0;
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_ALSA_SYM(x) \
if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1
#else
#define SDL_ALSA_SYM(x) ALSA_##x = x
#endif
static int
load_alsa_syms(void)
{
SDL_ALSA_SYM(snd_pcm_open);
SDL_ALSA_SYM(snd_pcm_close);
SDL_ALSA_SYM(snd_pcm_writei);
SDL_ALSA_SYM(snd_pcm_recover);
SDL_ALSA_SYM(snd_pcm_prepare);
SDL_ALSA_SYM(snd_pcm_drain);
SDL_ALSA_SYM(snd_strerror);
SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);
SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);
SDL_ALSA_SYM(snd_pcm_hw_params_copy);
SDL_ALSA_SYM(snd_pcm_hw_params_any);
SDL_ALSA_SYM(snd_pcm_hw_params_set_access);
SDL_ALSA_SYM(snd_pcm_hw_params_set_format);
SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);
SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);
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_get_periods);
SDL_ALSA_SYM(snd_pcm_hw_params_set_buffer_size_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size);
SDL_ALSA_SYM(snd_pcm_hw_params);
SDL_ALSA_SYM(snd_pcm_sw_params_current);
SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);
SDL_ALSA_SYM(snd_pcm_sw_params);
SDL_ALSA_SYM(snd_pcm_nonblock);
SDL_ALSA_SYM(snd_pcm_wait);
SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
return 0;
}
#undef SDL_ALSA_SYM
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static void
UnloadALSALibrary(void)
{
if (alsa_handle != NULL) {
SDL_UnloadObject(alsa_handle);
alsa_handle = NULL;
}
}
static int
LoadALSALibrary(void)
{
int retval = 0;
if (alsa_handle == NULL) {
alsa_handle = SDL_LoadObject(alsa_library);
if (alsa_handle == NULL) {
retval = -1;
/* Don't call SDL_SetError(): SDL_LoadObject already did. */
} else {
retval = load_alsa_syms();
if (retval < 0) {
UnloadALSALibrary();
}
}
}
return retval;
}
#else
static void
UnloadALSALibrary(void)
{
}
static int
LoadALSALibrary(void)
{
load_alsa_syms();
return 0;
}
#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
static const char *
get_audio_device(int channels)
{
const char *device;
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
if (device == NULL) {
switch (channels) {
case 6:
device = "plug:surround51";
break;
case 4:
device = "plug:surround40";
break;
default:
device = "default";
break;
}
}
return device;
}
/* This function waits until it is possible to write a full sound buffer */
static void
ALSA_WaitDevice(_THIS)
{
/* We're in blocking mode, so there's nothing to do here */
}
/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/*
* http://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
T *ptr = (T *) this->hidden->mixbuf; \
Uint32 i; \
for (i = 0; i < this->spec.samples; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
static SDL_INLINE void
swizzle_alsa_channels_6_64bit(_THIS)
{
SWIZ6(Uint64);
}
static SDL_INLINE void
swizzle_alsa_channels_6_32bit(_THIS)
{
SWIZ6(Uint32);
}
static SDL_INLINE void
swizzle_alsa_channels_6_16bit(_THIS)
{
SWIZ6(Uint16);
}
static SDL_INLINE void
swizzle_alsa_channels_6_8bit(_THIS)
{
SWIZ6(Uint8);
}
#undef SWIZ6
/*
* Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
* channels from Windows/Mac order to the format alsalib will want.
*/
static SDL_INLINE void
swizzle_alsa_channels(_THIS)
{
if (this->spec.channels == 6) {
const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
}
/* !!! FIXME: update this for 7.1 if needed, later. */
}
static void
ALSA_PlayDevice(_THIS)
{
int status;
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
swizzle_alsa_channels(this);
while ( frames_left > 0 && this->enabled ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
this->enabled = 0;
return;
}
continue;
}
sample_buf += status * frame_size;
frames_left -= status;
}
}
static Uint8 *
ALSA_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
ALSA_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
this->hidden->pcm_handle = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
{
int status;
snd_pcm_uframes_t bufsize;
/* "set" the hardware with the desired parameters */
status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
if ( status < 0 ) {
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 is useful for debugging */
if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {
snd_pcm_uframes_t persize = 0;
unsigned int periods = 0;
ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL);
ALSA_snd_pcm_hw_params_get_periods(hwparams, &periods, NULL);
fprintf(stderr,
"ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
persize, periods, bufsize);
}
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, const char *devname, int iscapture)
{
int status = 0;
snd_pcm_t *pcm_handle = NULL;
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_sw_params_t *swparams = NULL;
snd_pcm_format_t format = 0;
SDL_AudioFormat test_format = 0;
unsigned int rate = 0;
unsigned int channels = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status = ALSA_snd_pcm_open(&pcm_handle,
get_audio_device(this->spec.channels),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't open audio device: %s",
ALSA_snd_strerror(status));
}
this->hidden->pcm_handle = pcm_handle;
/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(&hwparams);
status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't get hardware config: %s",
ALSA_snd_strerror(status));
}
/* SDL only uses interleaved sample output */
status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set interleaved access: %s",
ALSA_snd_strerror(status));
}
/* Try for a closest match on audio format */
status = -1;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
test_format && (status < 0);) {
status = 0; /* if we can't support a format, it'll become -1. */
switch (test_format) {
case AUDIO_U8:
format = SND_PCM_FORMAT_U8;
break;
case AUDIO_S8:
format = SND_PCM_FORMAT_S8;
break;
case AUDIO_S16LSB:
format = SND_PCM_FORMAT_S16_LE;
break;
case AUDIO_S16MSB:
format = SND_PCM_FORMAT_S16_BE;
break;
case AUDIO_U16LSB:
format = SND_PCM_FORMAT_U16_LE;
break;
case AUDIO_U16MSB:
format = SND_PCM_FORMAT_U16_BE;
break;
case AUDIO_S32LSB:
format = SND_PCM_FORMAT_S32_LE;
break;
case AUDIO_S32MSB:
format = SND_PCM_FORMAT_S32_BE;
break;
case AUDIO_F32LSB:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AUDIO_F32MSB:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
default:
status = -1;
break;
}
if (status >= 0) {
status = ALSA_snd_pcm_hw_params_set_format(pcm_handle,
hwparams, format);
}
if (status < 0) {
test_format = SDL_NextAudioFormat();
}
}
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Set the number of channels */
status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
this->spec.channels);
channels = this->spec.channels;
if (status < 0) {
status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set audio channels");
}
this->spec.channels = channels;
}
/* Set the audio rate */
rate = this->spec.freq;
status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
&rate, NULL);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set audio frequency: %s",
ALSA_snd_strerror(status));
}
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... */
if ( ALSA_set_period_size(this, hwparams, 1) < 0 ) {
ALSA_CloseDevice(this);
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);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't get software config: %s",
ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set minimum available samples: %s",
ALSA_snd_strerror(status));
}
status =
ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set start threshold: %s",
ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set software audio parameters: %s",
ALSA_snd_strerror(status));
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ALSA_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
/* Switch to blocking mode for playback */
ALSA_snd_pcm_nonblock(pcm_handle, 0);
/* We're ready to rock and roll. :-) */
return 0;
}
static void
ALSA_Deinitialize(void)
{
UnloadALSALibrary();
}
static int
ALSA_Init(SDL_AudioDriverImpl * impl)
{
if (LoadALSALibrary() < 0) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = ALSA_OpenDevice;
impl->WaitDevice = ALSA_WaitDevice;
impl->GetDeviceBuf = ALSA_GetDeviceBuf;
impl->PlayDevice = ALSA_PlayDevice;
impl->CloseDevice = ALSA_CloseDevice;
impl->Deinitialize = ALSA_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */
return 1; /* this audio target is available. */
}
AudioBootStrap ALSA_bootstrap = {
"alsa", "ALSA PCM audio", ALSA_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_ALSA */
/* vi: set ts=4 sw=4 expandtab: */