From a855d6202536ff28e5aae4f22a0f31d8f5b325d0 Mon Sep 17 00:00:00 2001 From: Franklin Wei Date: Sat, 21 Jan 2017 15:18:31 -0500 Subject: Port of Duke Nukem 3D This ports Fabien Sanglard's Chocolate Duke to run on a version of SDL for Rockbox. Change-Id: I8f2c4c78af19de10c1633ed7bb7a997b43256dd9 --- apps/plugins/sdl/src/SDL.c | 350 ++ apps/plugins/sdl/src/SDL_error.c | 238 + apps/plugins/sdl/src/SDL_error_c.h | 58 + apps/plugins/sdl/src/SDL_fatal.c | 134 + apps/plugins/sdl/src/SDL_fatal.h | 28 + apps/plugins/sdl/src/audio/SDL_audio.c | 707 +++ apps/plugins/sdl/src/audio/SDL_audio_c.h | 35 + apps/plugins/sdl/src/audio/SDL_audiocvt.c | 1510 ++++++ apps/plugins/sdl/src/audio/SDL_audiodev.c | 179 + apps/plugins/sdl/src/audio/SDL_audiodev_c.h | 26 + apps/plugins/sdl/src/audio/SDL_audiomem.h | 25 + apps/plugins/sdl/src/audio/SDL_mixer.c | 264 + apps/plugins/sdl/src/audio/SDL_mixer_MMX.c | 207 + apps/plugins/sdl/src/audio/SDL_mixer_MMX.h | 17 + apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.c | 183 + apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.h | 38 + apps/plugins/sdl/src/audio/SDL_mixer_m68k.c | 210 + apps/plugins/sdl/src/audio/SDL_mixer_m68k.h | 36 + apps/plugins/sdl/src/audio/SDL_sysaudio.h | 189 + apps/plugins/sdl/src/audio/SDL_wave.c | 596 +++ apps/plugins/sdl/src/audio/SDL_wave.h | 62 + apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c | 619 +++ apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h | 48 + apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c | 362 ++ apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h | 60 + apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc | 225 + apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h | 39 + apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c | 404 ++ apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h | 58 + apps/plugins/sdl/src/audio/dart/SDL_dart.c | 441 ++ apps/plugins/sdl/src/audio/dart/SDL_dart.h | 63 + apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c | 246 + apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h | 41 + apps/plugins/sdl/src/audio/dc/aica.c | 271 ++ apps/plugins/sdl/src/audio/dc/aica.h | 40 + apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c | 186 + apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h | 41 + apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c | 455 ++ apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h | 59 + apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c | 242 + apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h | 45 + apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c | 340 ++ apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h | 53 + apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.c | 156 + apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.h | 40 + apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c | 323 ++ apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h | 57 + apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c | 291 ++ apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h | 45 + apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c | 496 ++ apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h | 50 + apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c | 215 + apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h | 121 + .../sdl/src/audio/mint/SDL_mintaudio_dma8.c | 357 ++ .../sdl/src/audio/mint/SDL_mintaudio_dma8.h | 85 + .../sdl/src/audio/mint/SDL_mintaudio_gsxb.c | 436 ++ .../sdl/src/audio/mint/SDL_mintaudio_gsxb.h | 104 + apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S | 386 ++ .../sdl/src/audio/mint/SDL_mintaudio_mcsn.c | 405 ++ .../sdl/src/audio/mint/SDL_mintaudio_mcsn.h | 59 + .../sdl/src/audio/mint/SDL_mintaudio_stfa.c | 326 ++ .../sdl/src/audio/mint/SDL_mintaudio_stfa.h | 97 + .../sdl/src/audio/mint/SDL_mintaudio_xbios.c | 490 ++ apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c | 264 + apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h | 51 + apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c | 423 ++ apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h | 62 + apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.c | 335 ++ apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.h | 40 + apps/plugins/sdl/src/audio/nds/sound9.c | 61 + apps/plugins/sdl/src/audio/nds/soundcommon.h | 80 + apps/plugins/sdl/src/audio/nto/SDL_nto_audio.c | 507 ++ apps/plugins/sdl/src/audio/nto/SDL_nto_audio.h | 68 + apps/plugins/sdl/src/audio/paudio/SDL_paudio.c | 511 ++ apps/plugins/sdl/src/audio/paudio/SDL_paudio.h | 57 + apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.c | 570 +++ apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.h | 73 + .../sdl/src/audio/rockbox/SDL_rockboxaudio.c | 271 ++ .../sdl/src/audio/rockbox/SDL_rockboxaudio.h | 50 + apps/plugins/sdl/src/audio/sun/SDL_sunaudio.c | 432 ++ apps/plugins/sdl/src/audio/sun/SDL_sunaudio.h | 55 + .../sdl/src/audio/symbian/SDL_epocaudio.cpp | 614 +++ apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.h | 37 + .../plugins/sdl/src/audio/symbian/streamplayer.cpp | 279 ++ apps/plugins/sdl/src/audio/symbian/streamplayer.h | 89 + apps/plugins/sdl/src/audio/ums/SDL_umsaudio.c | 547 +++ apps/plugins/sdl/src/audio/ums/SDL_umsaudio.h | 50 + apps/plugins/sdl/src/audio/windib/SDL_dibaudio.c | 322 ++ apps/plugins/sdl/src/audio/windib/SDL_dibaudio.h | 49 + apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.c | 705 +++ apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.h | 55 + apps/plugins/sdl/src/audio/windx5/directx.h | 81 + apps/plugins/sdl/src/cdrom/SDL_cdrom.c | 341 ++ apps/plugins/sdl/src/cdrom/SDL_syscdrom.h | 76 + apps/plugins/sdl/src/cdrom/aix/SDL_syscdrom.c | 660 +++ apps/plugins/sdl/src/cdrom/beos/SDL_syscdrom.cc | 410 ++ apps/plugins/sdl/src/cdrom/bsdi/SDL_syscdrom.c | 542 +++ apps/plugins/sdl/src/cdrom/dc/SDL_syscdrom.c | 167 + apps/plugins/sdl/src/cdrom/dummy/SDL_syscdrom.c | 41 + apps/plugins/sdl/src/cdrom/freebsd/SDL_syscdrom.c | 406 ++ apps/plugins/sdl/src/cdrom/linux/SDL_syscdrom.c | 564 +++ apps/plugins/sdl/src/cdrom/macos/SDL_syscdrom.c | 525 ++ apps/plugins/sdl/src/cdrom/macos/SDL_syscdrom_c.h | 140 + .../plugins/sdl/src/cdrom/macosx/AudioFilePlayer.c | 360 ++ .../plugins/sdl/src/cdrom/macosx/AudioFilePlayer.h | 178 + .../sdl/src/cdrom/macosx/AudioFileReaderThread.c | 610 +++ apps/plugins/sdl/src/cdrom/macosx/CDPlayer.c | 636 +++ apps/plugins/sdl/src/cdrom/macosx/CDPlayer.h | 69 + apps/plugins/sdl/src/cdrom/macosx/SDLOSXCAGuard.c | 199 + apps/plugins/sdl/src/cdrom/macosx/SDLOSXCAGuard.h | 116 + apps/plugins/sdl/src/cdrom/macosx/SDL_syscdrom.c | 514 ++ apps/plugins/sdl/src/cdrom/macosx/SDL_syscdrom_c.h | 136 + apps/plugins/sdl/src/cdrom/mint/SDL_syscdrom.c | 317 ++ apps/plugins/sdl/src/cdrom/openbsd/SDL_syscdrom.c | 416 ++ apps/plugins/sdl/src/cdrom/os2/SDL_syscdrom.c | 393 ++ apps/plugins/sdl/src/cdrom/osf/SDL_syscdrom.c | 444 ++ apps/plugins/sdl/src/cdrom/qnx/SDL_syscdrom.c | 551 +++ apps/plugins/sdl/src/cdrom/win32/SDL_syscdrom.c | 386 ++ apps/plugins/sdl/src/cpuinfo/SDL_cpuinfo.c | 499 ++ apps/plugins/sdl/src/events/SDL_active.c | 95 + apps/plugins/sdl/src/events/SDL_events.c | 501 ++ apps/plugins/sdl/src/events/SDL_events_c.h | 83 + apps/plugins/sdl/src/events/SDL_expose.c | 51 + apps/plugins/sdl/src/events/SDL_keyboard.c | 614 +++ apps/plugins/sdl/src/events/SDL_mouse.c | 268 + apps/plugins/sdl/src/events/SDL_quit.c | 124 + apps/plugins/sdl/src/events/SDL_resize.c | 71 + apps/plugins/sdl/src/events/SDL_sysevents.h | 46 + apps/plugins/sdl/src/file/SDL_rwops.c | 681 +++ apps/plugins/sdl/src/hermes/COPYING.LIB | 438 ++ apps/plugins/sdl/src/hermes/HeadMMX.h | 100 + apps/plugins/sdl/src/hermes/HeadX86.h | 186 + apps/plugins/sdl/src/hermes/README | 13 + apps/plugins/sdl/src/hermes/common.inc | 9 + apps/plugins/sdl/src/hermes/mmx_main.asm | 74 + apps/plugins/sdl/src/hermes/mmxp2_32.asm | 405 ++ apps/plugins/sdl/src/hermes/x86_main.asm | 75 + apps/plugins/sdl/src/hermes/x86p_16.asm | 490 ++ apps/plugins/sdl/src/hermes/x86p_32.asm | 1044 ++++ apps/plugins/sdl/src/joystick/SDL_joystick.c | 606 +++ apps/plugins/sdl/src/joystick/SDL_joystick_c.h | 38 + apps/plugins/sdl/src/joystick/SDL_sysjoystick.h | 82 + .../sdl/src/joystick/beos/SDL_bejoystick.cc | 237 + .../plugins/sdl/src/joystick/bsd/SDL_sysjoystick.c | 608 +++ .../sdl/src/joystick/darwin/SDL_sysjoystick.c | 842 ++++ apps/plugins/sdl/src/joystick/dc/SDL_sysjoystick.c | 193 + .../sdl/src/joystick/dummy/SDL_sysjoystick.c | 83 + .../sdl/src/joystick/linux/SDL_sysjoystick.c | 1218 +++++ .../sdl/src/joystick/macos/SDL_sysjoystick.c | 320 ++ .../sdl/src/joystick/mint/SDL_sysjoystick.c | 826 ++++ .../plugins/sdl/src/joystick/nds/SDL_sysjoystick.c | 150 + .../sdl/src/joystick/riscos/SDL_sysjoystick.c | 176 + .../sdl/src/joystick/win32/SDL_mmjoystick.c | 407 ++ apps/plugins/sdl/src/loadso/beos/SDL_sysloadso.c | 72 + apps/plugins/sdl/src/loadso/dlopen/SDL_sysloadso.c | 69 + apps/plugins/sdl/src/loadso/dummy/SDL_sysloadso.c | 50 + apps/plugins/sdl/src/loadso/macos/SDL_sysloadso.c | 106 + apps/plugins/sdl/src/loadso/macosx/SDL_dlcompat.c | 1407 ++++++ apps/plugins/sdl/src/loadso/mint/SDL_sysloadso.c | 62 + apps/plugins/sdl/src/loadso/os2/SDL_sysloadso.c | 71 + apps/plugins/sdl/src/loadso/win32/SDL_sysloadso.c | 139 + apps/plugins/sdl/src/main/beos/SDL_BeApp.cc | 111 + apps/plugins/sdl/src/main/beos/SDL_BeApp.h | 33 + apps/plugins/sdl/src/main/dummy/SDL_dummy_main.c | 13 + apps/plugins/sdl/src/main/macos/SDL.r | 1 + apps/plugins/sdl/src/main/macos/SDL.shlib.r | 1 + apps/plugins/sdl/src/main/macos/SDL_main.c | 610 +++ apps/plugins/sdl/src/main/macos/SIZE.r | 1 + apps/plugins/sdl/src/main/macos/exports/Makefile | 39 + apps/plugins/sdl/src/main/macos/exports/SDL.x | 1 + apps/plugins/sdl/src/main/macos/exports/gendef.pl | 43 + apps/plugins/sdl/src/main/macosx/Info.plist.in | 24 + apps/plugins/sdl/src/main/macosx/SDLMain.h | 16 + apps/plugins/sdl/src/main/macosx/SDLMain.m | 381 ++ .../sdl/src/main/macosx/SDLMain.nib/classes.nib | 12 + .../sdl/src/main/macosx/SDLMain.nib/info.nib | 12 + .../sdl/src/main/macosx/SDLMain.nib/objects.nib | Bin 0 -> 1701 bytes apps/plugins/sdl/src/main/macosx/info.nib | 1 + .../plugins/sdl/src/main/qtopia/SDL_qtopia_main.cc | 47 + .../plugins/sdl/src/main/symbian/EKA1/SDL_main.cpp | 152 + .../plugins/sdl/src/main/symbian/EKA2/SDL_main.cpp | 1035 ++++ apps/plugins/sdl/src/main/symbian/EKA2/sdlexe.cpp | 809 ++++ apps/plugins/sdl/src/main/symbian/EKA2/sdllib.cpp | 12 + .../sdl/src/main/symbian/EKA2/vectorbuffer.cpp | 62 + .../sdl/src/main/symbian/EKA2/vectorbuffer.h | 240 + apps/plugins/sdl/src/main/win32/SDL_win32_main.c | 402 ++ apps/plugins/sdl/src/main/win32/version.rc | 38 + apps/plugins/sdl/src/stdlib/SDL_getenv.c | 247 + apps/plugins/sdl/src/stdlib/SDL_iconv.c | 881 ++++ apps/plugins/sdl/src/stdlib/SDL_malloc.c | 5111 ++++++++++++++++++++ apps/plugins/sdl/src/stdlib/SDL_qsort.c | 443 ++ apps/plugins/sdl/src/stdlib/SDL_stdlib.c | 620 +++ apps/plugins/sdl/src/stdlib/SDL_string.c | 1248 +++++ apps/plugins/sdl/src/thread/SDL_systhread.h | 52 + apps/plugins/sdl/src/thread/SDL_thread.c | 300 ++ apps/plugins/sdl/src/thread/SDL_thread_c.h | 66 + apps/plugins/sdl/src/thread/beos/SDL_syssem.c | 142 + apps/plugins/sdl/src/thread/beos/SDL_systhread.c | 96 + apps/plugins/sdl/src/thread/beos/SDL_systhread_c.h | 31 + apps/plugins/sdl/src/thread/dc/SDL_syscond.c | 215 + apps/plugins/sdl/src/thread/dc/SDL_syscond_c.h | 23 + apps/plugins/sdl/src/thread/dc/SDL_sysmutex.c | 122 + apps/plugins/sdl/src/thread/dc/SDL_sysmutex_c.h | 23 + apps/plugins/sdl/src/thread/dc/SDL_syssem.c | 173 + apps/plugins/sdl/src/thread/dc/SDL_syssem_c.h | 23 + apps/plugins/sdl/src/thread/dc/SDL_systhread.c | 60 + apps/plugins/sdl/src/thread/dc/SDL_systhread_c.h | 24 + apps/plugins/sdl/src/thread/generic/SDL_syscond.c | 215 + apps/plugins/sdl/src/thread/generic/SDL_sysmutex.c | 129 + .../sdl/src/thread/generic/SDL_sysmutex_c.h | 23 + apps/plugins/sdl/src/thread/generic/SDL_syssem.c | 211 + .../plugins/sdl/src/thread/generic/SDL_systhread.c | 54 + .../sdl/src/thread/generic/SDL_systhread_c.h | 25 + apps/plugins/sdl/src/thread/irix/SDL_syssem.c | 219 + apps/plugins/sdl/src/thread/irix/SDL_systhread.c | 85 + apps/plugins/sdl/src/thread/irix/SDL_systhread_c.h | 27 + apps/plugins/sdl/src/thread/os2/SDL_syscond.c | 215 + apps/plugins/sdl/src/thread/os2/SDL_syscond_c.h | 23 + apps/plugins/sdl/src/thread/os2/SDL_sysmutex.c | 108 + apps/plugins/sdl/src/thread/os2/SDL_syssem.c | 192 + apps/plugins/sdl/src/thread/os2/SDL_systhread.c | 108 + apps/plugins/sdl/src/thread/os2/SDL_systhread_c.h | 28 + apps/plugins/sdl/src/thread/pth/SDL_syscond.c | 164 + apps/plugins/sdl/src/thread/pth/SDL_sysmutex.c | 87 + apps/plugins/sdl/src/thread/pth/SDL_sysmutex_c.h | 31 + apps/plugins/sdl/src/thread/pth/SDL_systhread.c | 103 + apps/plugins/sdl/src/thread/pth/SDL_systhread_c.h | 31 + apps/plugins/sdl/src/thread/pthread/SDL_syscond.c | 155 + apps/plugins/sdl/src/thread/pthread/SDL_sysmutex.c | 153 + .../sdl/src/thread/pthread/SDL_sysmutex_c.h | 31 + apps/plugins/sdl/src/thread/pthread/SDL_syssem.c | 190 + .../plugins/sdl/src/thread/pthread/SDL_systhread.c | 120 + .../sdl/src/thread/pthread/SDL_systhread_c.h | 26 + apps/plugins/sdl/src/thread/riscos/SDL_syscond.c | 160 + apps/plugins/sdl/src/thread/riscos/SDL_sysmutex.c | 153 + .../plugins/sdl/src/thread/riscos/SDL_sysmutex_c.h | 34 + apps/plugins/sdl/src/thread/riscos/SDL_syssem.c | 203 + apps/plugins/sdl/src/thread/riscos/SDL_systhread.c | 144 + .../sdl/src/thread/riscos/SDL_systhread_c.h | 34 + apps/plugins/sdl/src/thread/rockbox/SDL_syscond.c | 215 + apps/plugins/sdl/src/thread/rockbox/SDL_sysmutex.c | 120 + .../sdl/src/thread/rockbox/SDL_sysmutex_c.h | 23 + apps/plugins/sdl/src/thread/rockbox/SDL_syssem.c | 170 + .../plugins/sdl/src/thread/rockbox/SDL_systhread.c | 89 + .../sdl/src/thread/rockbox/SDL_systhread_c.h | 24 + .../sdl/src/thread/symbian/SDL_sysmutex.cpp | 130 + apps/plugins/sdl/src/thread/symbian/SDL_syssem.cpp | 214 + .../sdl/src/thread/symbian/SDL_systhread.cpp | 146 + .../sdl/src/thread/symbian/SDL_systhread_c.h | 30 + apps/plugins/sdl/src/thread/win32/SDL_sysmutex.c | 95 + apps/plugins/sdl/src/thread/win32/SDL_syssem.c | 164 + apps/plugins/sdl/src/thread/win32/SDL_systhread.c | 162 + .../plugins/sdl/src/thread/win32/SDL_systhread_c.h | 28 + .../sdl/src/thread/win32/win_ce_semaphore.c | 216 + .../sdl/src/thread/win32/win_ce_semaphore.h | 22 + apps/plugins/sdl/src/timer/SDL_systimer.h | 40 + apps/plugins/sdl/src/timer/SDL_timer.c | 285 ++ apps/plugins/sdl/src/timer/SDL_timer_c.h | 46 + apps/plugins/sdl/src/timer/beos/SDL_systimer.c | 95 + apps/plugins/sdl/src/timer/dc/SDL_systimer.c | 100 + apps/plugins/sdl/src/timer/dummy/SDL_systimer.c | 91 + apps/plugins/sdl/src/timer/macos/FastTimes.c | 352 ++ apps/plugins/sdl/src/timer/macos/FastTimes.h | 27 + apps/plugins/sdl/src/timer/macos/SDL_MPWtimer.c | 152 + apps/plugins/sdl/src/timer/macos/SDL_systimer.c | 186 + apps/plugins/sdl/src/timer/mint/SDL_systimer.c | 149 + apps/plugins/sdl/src/timer/mint/SDL_vbltimer.S | 228 + apps/plugins/sdl/src/timer/mint/SDL_vbltimer_s.h | 35 + apps/plugins/sdl/src/timer/nds/SDL_systimer.c | 73 + apps/plugins/sdl/src/timer/os2/SDL_systimer.c | 227 + apps/plugins/sdl/src/timer/riscos/SDL_systimer.c | 233 + apps/plugins/sdl/src/timer/rockbox/SDL_systimer.c | 117 + .../plugins/sdl/src/timer/symbian/SDL_systimer.cpp | 114 + apps/plugins/sdl/src/timer/unix/SDL_systimer.c | 240 + apps/plugins/sdl/src/timer/win32/SDL_systimer.c | 160 + apps/plugins/sdl/src/timer/wince/SDL_systimer.c | 198 + apps/plugins/sdl/src/video/SDL_RLEaccel.c | 1941 ++++++++ apps/plugins/sdl/src/video/SDL_RLEaccel_c.h | 31 + apps/plugins/sdl/src/video/SDL_blit.c | 360 ++ apps/plugins/sdl/src/video/SDL_blit.h | 528 ++ apps/plugins/sdl/src/video/SDL_blit_0.c | 471 ++ apps/plugins/sdl/src/video/SDL_blit_1.c | 523 ++ apps/plugins/sdl/src/video/SDL_blit_A.c | 2873 +++++++++++ apps/plugins/sdl/src/video/SDL_blit_N.c | 2492 ++++++++++ apps/plugins/sdl/src/video/SDL_bmp.c | 549 +++ apps/plugins/sdl/src/video/SDL_cursor.c | 758 +++ apps/plugins/sdl/src/video/SDL_cursor_c.h | 73 + apps/plugins/sdl/src/video/SDL_gamma.c | 233 + apps/plugins/sdl/src/video/SDL_glfuncs.h | 341 ++ apps/plugins/sdl/src/video/SDL_leaks.h | 31 + apps/plugins/sdl/src/video/SDL_pixels.c | 626 +++ apps/plugins/sdl/src/video/SDL_pixels_c.h | 46 + apps/plugins/sdl/src/video/SDL_stretch.c | 358 ++ apps/plugins/sdl/src/video/SDL_stretch_c.h | 29 + apps/plugins/sdl/src/video/SDL_surface.c | 941 ++++ apps/plugins/sdl/src/video/SDL_sysvideo.h | 427 ++ apps/plugins/sdl/src/video/SDL_video.c | 1978 ++++++++ apps/plugins/sdl/src/video/SDL_yuv.c | 150 + apps/plugins/sdl/src/video/SDL_yuv_mmx.c | 428 ++ apps/plugins/sdl/src/video/SDL_yuv_sw.c | 1299 +++++ apps/plugins/sdl/src/video/SDL_yuv_sw_c.h | 37 + apps/plugins/sdl/src/video/SDL_yuvfuncs.h | 37 + apps/plugins/sdl/src/video/Xext/README | 10 + apps/plugins/sdl/src/video/Xext/XME/xme.c | 410 ++ .../plugins/sdl/src/video/Xext/Xinerama/Xinerama.c | 324 ++ apps/plugins/sdl/src/video/Xext/Xv/Xv.c | 1151 +++++ apps/plugins/sdl/src/video/Xext/Xv/Xvlibint.h | 81 + apps/plugins/sdl/src/video/Xext/Xxf86dga/XF86DGA.c | 721 +++ .../plugins/sdl/src/video/Xext/Xxf86dga/XF86DGA2.c | 993 ++++ .../plugins/sdl/src/video/Xext/Xxf86vm/XF86VMode.c | 1226 +++++ apps/plugins/sdl/src/video/Xext/extensions/Xext.h | 50 + .../sdl/src/video/Xext/extensions/Xinerama.h | 46 + apps/plugins/sdl/src/video/Xext/extensions/Xv.h | 129 + apps/plugins/sdl/src/video/Xext/extensions/Xvlib.h | 433 ++ .../sdl/src/video/Xext/extensions/Xvproto.h | 604 +++ .../sdl/src/video/Xext/extensions/extutil.h | 226 + .../sdl/src/video/Xext/extensions/panoramiXext.h | 52 + .../sdl/src/video/Xext/extensions/panoramiXproto.h | 192 + .../sdl/src/video/Xext/extensions/xf86dga.h | 265 + .../sdl/src/video/Xext/extensions/xf86dga1.h | 169 + .../sdl/src/video/Xext/extensions/xf86dga1str.h | 194 + .../sdl/src/video/Xext/extensions/xf86dgastr.h | 344 ++ .../sdl/src/video/Xext/extensions/xf86vmode.h | 314 ++ .../sdl/src/video/Xext/extensions/xf86vmstr.h | 546 +++ apps/plugins/sdl/src/video/Xext/extensions/xme.h | 45 + apps/plugins/sdl/src/video/aalib/SDL_aaevents.c | 202 + apps/plugins/sdl/src/video/aalib/SDL_aaevents_c.h | 35 + apps/plugins/sdl/src/video/aalib/SDL_aamouse.c | 35 + apps/plugins/sdl/src/video/aalib/SDL_aamouse_c.h | 26 + apps/plugins/sdl/src/video/aalib/SDL_aavideo.c | 388 ++ apps/plugins/sdl/src/video/aalib/SDL_aavideo.h | 66 + .../sdl/src/video/ataricommon/SDL_ataric2p.S | 452 ++ .../sdl/src/video/ataricommon/SDL_ataric2p_s.h | 75 + .../sdl/src/video/ataricommon/SDL_ataridevmouse.c | 159 + .../src/video/ataricommon/SDL_ataridevmouse_c.h | 42 + .../sdl/src/video/ataricommon/SDL_atarieddi.S | 42 + .../sdl/src/video/ataricommon/SDL_atarieddi_s.h | 54 + .../sdl/src/video/ataricommon/SDL_atarievents.c | 234 + .../sdl/src/video/ataricommon/SDL_atarievents_c.h | 52 + .../sdl/src/video/ataricommon/SDL_atarigl.c | 1086 +++++ .../sdl/src/video/ataricommon/SDL_atarigl_c.h | 109 + .../sdl/src/video/ataricommon/SDL_atarikeys.h | 140 + .../sdl/src/video/ataricommon/SDL_atarimxalloc.c | 52 + .../sdl/src/video/ataricommon/SDL_atarimxalloc_c.h | 37 + .../sdl/src/video/ataricommon/SDL_atarisuper.h | 61 + .../sdl/src/video/ataricommon/SDL_biosevents.c | 131 + .../sdl/src/video/ataricommon/SDL_biosevents_c.h | 42 + .../sdl/src/video/ataricommon/SDL_gemdosevents.c | 132 + .../sdl/src/video/ataricommon/SDL_gemdosevents_c.h | 42 + .../sdl/src/video/ataricommon/SDL_ikbdevents.c | 124 + .../sdl/src/video/ataricommon/SDL_ikbdevents_c.h | 42 + .../sdl/src/video/ataricommon/SDL_ikbdinterrupt.S | 404 ++ .../src/video/ataricommon/SDL_ikbdinterrupt_s.h | 61 + .../sdl/src/video/ataricommon/SDL_xbiosevents.c | 155 + .../sdl/src/video/ataricommon/SDL_xbiosevents_c.h | 48 + .../sdl/src/video/ataricommon/SDL_xbiosinterrupt.S | 212 + .../src/video/ataricommon/SDL_xbiosinterrupt_s.h | 52 + apps/plugins/sdl/src/video/blank_cursor.h | 33 + apps/plugins/sdl/src/video/bwindow/SDL_BView.h | 116 + apps/plugins/sdl/src/video/bwindow/SDL_BWin.h | 290 ++ apps/plugins/sdl/src/video/bwindow/SDL_lowvideo.h | 58 + .../plugins/sdl/src/video/bwindow/SDL_sysevents.cc | 415 ++ .../sdl/src/video/bwindow/SDL_sysevents_c.h | 31 + apps/plugins/sdl/src/video/bwindow/SDL_sysmouse.cc | 153 + .../plugins/sdl/src/video/bwindow/SDL_sysmouse_c.h | 33 + apps/plugins/sdl/src/video/bwindow/SDL_sysvideo.cc | 841 ++++ apps/plugins/sdl/src/video/bwindow/SDL_syswm.cc | 92 + apps/plugins/sdl/src/video/bwindow/SDL_syswm_c.h | 32 + apps/plugins/sdl/src/video/bwindow/SDL_sysyuv.cc | 314 ++ apps/plugins/sdl/src/video/bwindow/SDL_sysyuv.h | 73 + apps/plugins/sdl/src/video/caca/SDL_cacaevents.c | 101 + apps/plugins/sdl/src/video/caca/SDL_cacaevents_c.h | 35 + apps/plugins/sdl/src/video/caca/SDL_cacavideo.c | 304 ++ apps/plugins/sdl/src/video/caca/SDL_cacavideo.h | 76 + apps/plugins/sdl/src/video/dc/SDL_dcevents.c | 152 + apps/plugins/sdl/src/video/dc/SDL_dcevents_c.h | 33 + apps/plugins/sdl/src/video/dc/SDL_dcmouse.c | 35 + apps/plugins/sdl/src/video/dc/SDL_dcmouse_c.h | 26 + apps/plugins/sdl/src/video/dc/SDL_dcvideo.c | 445 ++ apps/plugins/sdl/src/video/dc/SDL_dcvideo.h | 42 + apps/plugins/sdl/src/video/default_cursor.h | 116 + apps/plugins/sdl/src/video/dga/SDL_dgaevents.c | 163 + apps/plugins/sdl/src/video/dga/SDL_dgaevents_c.h | 28 + apps/plugins/sdl/src/video/dga/SDL_dgamouse.c | 35 + apps/plugins/sdl/src/video/dga/SDL_dgamouse_c.h | 26 + apps/plugins/sdl/src/video/dga/SDL_dgavideo.c | 1101 +++++ apps/plugins/sdl/src/video/dga/SDL_dgavideo.h | 124 + .../sdl/src/video/directfb/SDL_DirectFB_events.c | 219 + .../sdl/src/video/directfb/SDL_DirectFB_events.h | 29 + .../sdl/src/video/directfb/SDL_DirectFB_keys.h | 135 + .../sdl/src/video/directfb/SDL_DirectFB_video.c | 1171 +++++ .../sdl/src/video/directfb/SDL_DirectFB_video.h | 62 + .../sdl/src/video/directfb/SDL_DirectFB_yuv.c | 290 ++ .../sdl/src/video/directfb/SDL_DirectFB_yuv.h | 38 + apps/plugins/sdl/src/video/dummy/SDL_nullevents.c | 45 + .../plugins/sdl/src/video/dummy/SDL_nullevents_c.h | 33 + apps/plugins/sdl/src/video/dummy/SDL_nullmouse.c | 33 + apps/plugins/sdl/src/video/dummy/SDL_nullmouse_c.h | 26 + apps/plugins/sdl/src/video/dummy/SDL_nullvideo.c | 239 + apps/plugins/sdl/src/video/dummy/SDL_nullvideo.h | 40 + apps/plugins/sdl/src/video/e_log.h | 140 + apps/plugins/sdl/src/video/e_pow.h | 302 ++ apps/plugins/sdl/src/video/e_sqrt.h | 493 ++ apps/plugins/sdl/src/video/fbcon/3dfx_mmio.h | 56 + apps/plugins/sdl/src/video/fbcon/3dfx_regs.h | 83 + apps/plugins/sdl/src/video/fbcon/SDL_fb3dfx.c | 215 + apps/plugins/sdl/src/video/fbcon/SDL_fb3dfx.h | 29 + apps/plugins/sdl/src/video/fbcon/SDL_fbelo.c | 442 ++ apps/plugins/sdl/src/video/fbcon/SDL_fbelo.h | 55 + apps/plugins/sdl/src/video/fbcon/SDL_fbevents.c | 1254 +++++ apps/plugins/sdl/src/video/fbcon/SDL_fbevents_c.h | 38 + apps/plugins/sdl/src/video/fbcon/SDL_fbkeys.h | 139 + apps/plugins/sdl/src/video/fbcon/SDL_fbmatrox.c | 280 ++ apps/plugins/sdl/src/video/fbcon/SDL_fbmatrox.h | 29 + apps/plugins/sdl/src/video/fbcon/SDL_fbmouse.c | 33 + apps/plugins/sdl/src/video/fbcon/SDL_fbmouse_c.h | 26 + apps/plugins/sdl/src/video/fbcon/SDL_fbriva.c | 222 + apps/plugins/sdl/src/video/fbcon/SDL_fbriva.h | 36 + apps/plugins/sdl/src/video/fbcon/SDL_fbvideo.c | 1982 ++++++++ apps/plugins/sdl/src/video/fbcon/SDL_fbvideo.h | 200 + apps/plugins/sdl/src/video/fbcon/matrox_mmio.h | 51 + apps/plugins/sdl/src/video/fbcon/matrox_regs.h | 376 ++ apps/plugins/sdl/src/video/fbcon/riva_mmio.h | 449 ++ apps/plugins/sdl/src/video/fbcon/riva_regs.h | 43 + apps/plugins/sdl/src/video/gapi/SDL_gapivideo.c | 1287 +++++ apps/plugins/sdl/src/video/gapi/SDL_gapivideo.h | 160 + apps/plugins/sdl/src/video/gem/SDL_gemevents.c | 375 ++ apps/plugins/sdl/src/video/gem/SDL_gemevents_c.h | 33 + apps/plugins/sdl/src/video/gem/SDL_gemmouse.c | 205 + apps/plugins/sdl/src/video/gem/SDL_gemmouse_c.h | 34 + apps/plugins/sdl/src/video/gem/SDL_gemvideo.c | 1337 +++++ apps/plugins/sdl/src/video/gem/SDL_gemvideo.h | 191 + apps/plugins/sdl/src/video/gem/SDL_gemwm.c | 116 + apps/plugins/sdl/src/video/gem/SDL_gemwm_c.h | 37 + apps/plugins/sdl/src/video/ggi/SDL_ggievents.c | 264 + apps/plugins/sdl/src/video/ggi/SDL_ggievents_c.h | 29 + apps/plugins/sdl/src/video/ggi/SDL_ggikeys.h | 135 + apps/plugins/sdl/src/video/ggi/SDL_ggimouse.c | 32 + apps/plugins/sdl/src/video/ggi/SDL_ggimouse_c.h | 26 + apps/plugins/sdl/src/video/ggi/SDL_ggivideo.c | 378 ++ apps/plugins/sdl/src/video/ggi/SDL_ggivideo.h | 48 + apps/plugins/sdl/src/video/ipod/SDL_ipodvideo.c | 733 +++ apps/plugins/sdl/src/video/ipod/SDL_ipodvideo.h | 38 + .../plugins/sdl/src/video/maccommon/SDL_lowvideo.h | 102 + .../sdl/src/video/maccommon/SDL_macevents.c | 746 +++ .../sdl/src/video/maccommon/SDL_macevents_c.h | 32 + apps/plugins/sdl/src/video/maccommon/SDL_macgl.c | 197 + apps/plugins/sdl/src/video/maccommon/SDL_macgl_c.h | 47 + apps/plugins/sdl/src/video/maccommon/SDL_mackeys.h | 140 + .../plugins/sdl/src/video/maccommon/SDL_macmouse.c | 129 + .../sdl/src/video/maccommon/SDL_macmouse_c.h | 34 + apps/plugins/sdl/src/video/maccommon/SDL_macwm.c | 442 ++ apps/plugins/sdl/src/video/maccommon/SDL_macwm_c.h | 41 + apps/plugins/sdl/src/video/macdsp/SDL_dspvideo.c | 1422 ++++++ apps/plugins/sdl/src/video/macdsp/SDL_dspvideo.h | 54 + apps/plugins/sdl/src/video/macrom/SDL_romvideo.c | 745 +++ apps/plugins/sdl/src/video/macrom/SDL_romvideo.h | 29 + apps/plugins/sdl/src/video/math_private.h | 173 + apps/plugins/sdl/src/video/mmx.h | 704 +++ apps/plugins/sdl/src/video/nanox/SDL_nxevents.c | 382 ++ apps/plugins/sdl/src/video/nanox/SDL_nxevents_c.h | 32 + apps/plugins/sdl/src/video/nanox/SDL_nximage.c | 230 + apps/plugins/sdl/src/video/nanox/SDL_nximage_c.h | 35 + apps/plugins/sdl/src/video/nanox/SDL_nxmodes.c | 84 + apps/plugins/sdl/src/video/nanox/SDL_nxmodes_c.h | 34 + apps/plugins/sdl/src/video/nanox/SDL_nxmouse.c | 79 + apps/plugins/sdl/src/video/nanox/SDL_nxmouse_c.h | 29 + apps/plugins/sdl/src/video/nanox/SDL_nxvideo.c | 544 +++ apps/plugins/sdl/src/video/nanox/SDL_nxvideo.h | 96 + apps/plugins/sdl/src/video/nanox/SDL_nxwm.c | 61 + apps/plugins/sdl/src/video/nanox/SDL_nxwm_c.h | 32 + apps/plugins/sdl/src/video/nds/SDL_ndsevents.c | 83 + apps/plugins/sdl/src/video/nds/SDL_ndsevents_c.h | 51 + apps/plugins/sdl/src/video/nds/SDL_ndsmouse.c | 34 + apps/plugins/sdl/src/video/nds/SDL_ndsmouse_c.h | 26 + apps/plugins/sdl/src/video/nds/SDL_ndsvideo.c | 500 ++ apps/plugins/sdl/src/video/nds/SDL_ndsvideo.h | 61 + apps/plugins/sdl/src/video/os2fslib/SDL_os2fslib.c | 3018 ++++++++++++ apps/plugins/sdl/src/video/os2fslib/SDL_os2fslib.h | 71 + apps/plugins/sdl/src/video/os2fslib/SDL_vkeys.h | 74 + apps/plugins/sdl/src/video/photon/SDL_ph_events.c | 624 +++ .../plugins/sdl/src/video/photon/SDL_ph_events_c.h | 37 + apps/plugins/sdl/src/video/photon/SDL_ph_gl.c | 406 ++ apps/plugins/sdl/src/video/photon/SDL_ph_gl.h | 41 + apps/plugins/sdl/src/video/photon/SDL_ph_image.c | 1059 ++++ apps/plugins/sdl/src/video/photon/SDL_ph_image_c.h | 59 + apps/plugins/sdl/src/video/photon/SDL_ph_modes.c | 390 ++ apps/plugins/sdl/src/video/photon/SDL_ph_modes_c.h | 43 + apps/plugins/sdl/src/video/photon/SDL_ph_mouse.c | 220 + apps/plugins/sdl/src/video/photon/SDL_ph_mouse_c.h | 39 + apps/plugins/sdl/src/video/photon/SDL_ph_video.c | 648 +++ apps/plugins/sdl/src/video/photon/SDL_ph_video.h | 157 + apps/plugins/sdl/src/video/photon/SDL_ph_wm.c | 118 + apps/plugins/sdl/src/video/photon/SDL_ph_wm_c.h | 37 + apps/plugins/sdl/src/video/photon/SDL_phyuv.c | 504 ++ apps/plugins/sdl/src/video/photon/SDL_phyuv_c.h | 62 + apps/plugins/sdl/src/video/picogui/SDL_pgevents.c | 117 + .../plugins/sdl/src/video/picogui/SDL_pgevents_c.h | 37 + apps/plugins/sdl/src/video/picogui/SDL_pgvideo.c | 364 ++ apps/plugins/sdl/src/video/picogui/SDL_pgvideo.h | 50 + apps/plugins/sdl/src/video/ps2gs/SDL_gsevents.c | 977 ++++ apps/plugins/sdl/src/video/ps2gs/SDL_gsevents_c.h | 38 + apps/plugins/sdl/src/video/ps2gs/SDL_gskeys.h | 139 + apps/plugins/sdl/src/video/ps2gs/SDL_gsmouse.c | 146 + apps/plugins/sdl/src/video/ps2gs/SDL_gsmouse_c.h | 37 + apps/plugins/sdl/src/video/ps2gs/SDL_gsvideo.c | 689 +++ apps/plugins/sdl/src/video/ps2gs/SDL_gsvideo.h | 95 + apps/plugins/sdl/src/video/ps2gs/SDL_gsyuv.c | 461 ++ apps/plugins/sdl/src/video/ps2gs/SDL_gsyuv_c.h | 37 + apps/plugins/sdl/src/video/ps3/SDL_ps3events.c | 44 + apps/plugins/sdl/src/video/ps3/SDL_ps3events_c.h | 41 + apps/plugins/sdl/src/video/ps3/SDL_ps3video.c | 621 +++ apps/plugins/sdl/src/video/ps3/SDL_ps3video.h | 165 + apps/plugins/sdl/src/video/ps3/SDL_ps3yuv.c | 340 ++ apps/plugins/sdl/src/video/ps3/SDL_ps3yuv_c.h | 44 + apps/plugins/sdl/src/video/ps3/spulibs/Makefile | 83 + .../sdl/src/video/ps3/spulibs/bilin_scaler.c | 2050 ++++++++ apps/plugins/sdl/src/video/ps3/spulibs/fb_writer.c | 193 + .../plugins/sdl/src/video/ps3/spulibs/spu_common.h | 108 + .../sdl/src/video/ps3/spulibs/yuv2rgb_converter.c | 629 +++ apps/plugins/sdl/src/video/qtopia/SDL_QPEApp.cc | 63 + apps/plugins/sdl/src/video/qtopia/SDL_QPEApp.h | 33 + apps/plugins/sdl/src/video/qtopia/SDL_QWin.cc | 527 ++ apps/plugins/sdl/src/video/qtopia/SDL_QWin.h | 110 + apps/plugins/sdl/src/video/qtopia/SDL_lowvideo.h | 65 + apps/plugins/sdl/src/video/qtopia/SDL_sysevents.cc | 269 ++ .../plugins/sdl/src/video/qtopia/SDL_sysevents_c.h | 31 + apps/plugins/sdl/src/video/qtopia/SDL_sysmouse.cc | 56 + apps/plugins/sdl/src/video/qtopia/SDL_sysmouse_c.h | 32 + apps/plugins/sdl/src/video/qtopia/SDL_sysvideo.cc | 403 ++ apps/plugins/sdl/src/video/qtopia/SDL_syswm.cc | 35 + apps/plugins/sdl/src/video/qtopia/SDL_syswm_c.h | 28 + apps/plugins/sdl/src/video/quartz/CGS.h | 84 + .../sdl/src/video/quartz/SDL_QuartzEvents.m | 1063 ++++ apps/plugins/sdl/src/video/quartz/SDL_QuartzGL.m | 292 ++ apps/plugins/sdl/src/video/quartz/SDL_QuartzKeys.h | 146 + .../plugins/sdl/src/video/quartz/SDL_QuartzVideo.h | 229 + .../plugins/sdl/src/video/quartz/SDL_QuartzVideo.m | 1689 +++++++ apps/plugins/sdl/src/video/quartz/SDL_QuartzWM.h | 27 + apps/plugins/sdl/src/video/quartz/SDL_QuartzWM.m | 444 ++ .../sdl/src/video/quartz/SDL_QuartzWindow.h | 51 + .../sdl/src/video/quartz/SDL_QuartzWindow.m | 231 + apps/plugins/sdl/src/video/riscos/SDL_riscosASM.S | 116 + .../src/video/riscos/SDL_riscosFullScreenVideo.c | 777 +++ .../sdl/src/video/riscos/SDL_riscosevents.c | 549 +++ .../sdl/src/video/riscos/SDL_riscosevents_c.h | 34 + .../plugins/sdl/src/video/riscos/SDL_riscosmouse.c | 371 ++ .../sdl/src/video/riscos/SDL_riscosmouse_c.h | 44 + .../sdl/src/video/riscos/SDL_riscossprite.c | 265 + apps/plugins/sdl/src/video/riscos/SDL_riscostask.c | 350 ++ apps/plugins/sdl/src/video/riscos/SDL_riscostask.h | 39 + .../plugins/sdl/src/video/riscos/SDL_riscosvideo.c | 316 ++ .../plugins/sdl/src/video/riscos/SDL_riscosvideo.h | 62 + apps/plugins/sdl/src/video/riscos/SDL_wimppoll.c | 330 ++ apps/plugins/sdl/src/video/riscos/SDL_wimpvideo.c | 501 ++ .../sdl/src/video/rockbox/SDL_rockboxvideo.c | 793 +++ .../sdl/src/video/rockbox/SDL_rockboxvideo.h | 44 + apps/plugins/sdl/src/video/rockbox/keymaps_extra.h | 40 + apps/plugins/sdl/src/video/svga/SDL_svgaevents.c | 412 ++ apps/plugins/sdl/src/video/svga/SDL_svgaevents_c.h | 35 + apps/plugins/sdl/src/video/svga/SDL_svgamouse.c | 33 + apps/plugins/sdl/src/video/svga/SDL_svgamouse_c.h | 26 + apps/plugins/sdl/src/video/svga/SDL_svgavideo.c | 584 +++ apps/plugins/sdl/src/video/svga/SDL_svgavideo.h | 58 + .../sdl/src/video/symbian/EKA1/SDL_epocevents.cpp | 626 +++ .../sdl/src/video/symbian/EKA1/SDL_epocvideo.cpp | 1356 ++++++ .../sdl/src/video/symbian/EKA1/SDL_epocvideo.h | 34 + .../sdl/src/video/symbian/EKA2/SDL_epocevents.cpp | 521 ++ .../sdl/src/video/symbian/EKA2/SDL_epocvideo.cpp | 594 +++ .../sdl/src/video/symbian/EKA2/SDL_epocvideo.h | 51 + apps/plugins/sdl/src/video/symbian/EKA2/dsa.cpp | 1505 ++++++ .../plugins/sdl/src/video/symbian/EKA2/dsa_new.cpp | 1443 ++++++ .../plugins/sdl/src/video/symbian/EKA2/dsa_old.cpp | 1075 ++++ .../sdl/src/video/symbian/SDL_epocevents_c.h | 60 + apps/plugins/sdl/src/video/vgl/SDL_vglevents.c | 299 ++ apps/plugins/sdl/src/video/vgl/SDL_vglevents_c.h | 155 + apps/plugins/sdl/src/video/vgl/SDL_vglmouse.c | 56 + apps/plugins/sdl/src/video/vgl/SDL_vglmouse_c.h | 32 + apps/plugins/sdl/src/video/vgl/SDL_vglvideo.c | 624 +++ apps/plugins/sdl/src/video/vgl/SDL_vglvideo.h | 65 + .../plugins/sdl/src/video/wincommon/SDL_lowvideo.h | 152 + .../sdl/src/video/wincommon/SDL_sysevents.c | 855 ++++ .../plugins/sdl/src/video/wincommon/SDL_sysmouse.c | 259 + .../sdl/src/video/wincommon/SDL_sysmouse_c.h | 33 + apps/plugins/sdl/src/video/wincommon/SDL_syswm.c | 297 ++ apps/plugins/sdl/src/video/wincommon/SDL_syswm_c.h | 35 + apps/plugins/sdl/src/video/wincommon/SDL_wingl.c | 659 +++ apps/plugins/sdl/src/video/wincommon/SDL_wingl_c.h | 135 + apps/plugins/sdl/src/video/wincommon/wmmsg.h | 1030 ++++ apps/plugins/sdl/src/video/windib/SDL_dibevents.c | 704 +++ .../plugins/sdl/src/video/windib/SDL_dibevents_c.h | 35 + apps/plugins/sdl/src/video/windib/SDL_dibvideo.c | 1323 +++++ apps/plugins/sdl/src/video/windib/SDL_dibvideo.h | 59 + .../sdl/src/video/windib/SDL_gapidibvideo.h | 56 + apps/plugins/sdl/src/video/windib/SDL_vkeys.h | 75 + apps/plugins/sdl/src/video/windx5/SDL_dx5events.c | 1005 ++++ .../plugins/sdl/src/video/windx5/SDL_dx5events_c.h | 37 + apps/plugins/sdl/src/video/windx5/SDL_dx5video.c | 2537 ++++++++++ apps/plugins/sdl/src/video/windx5/SDL_dx5video.h | 61 + apps/plugins/sdl/src/video/windx5/SDL_dx5yuv.c | 296 ++ apps/plugins/sdl/src/video/windx5/SDL_dx5yuv_c.h | 38 + apps/plugins/sdl/src/video/windx5/directx.h | 97 + .../sdl/src/video/wscons/SDL_wsconsevents.c | 233 + .../sdl/src/video/wscons/SDL_wsconsevents_c.h | 36 + .../plugins/sdl/src/video/wscons/SDL_wsconsmouse.c | 33 + .../sdl/src/video/wscons/SDL_wsconsmouse_c.h | 26 + .../plugins/sdl/src/video/wscons/SDL_wsconsvideo.c | 609 +++ .../plugins/sdl/src/video/wscons/SDL_wsconsvideo.h | 76 + apps/plugins/sdl/src/video/x11/SDL_x11dga.c | 90 + apps/plugins/sdl/src/video/x11/SDL_x11dga_c.h | 33 + apps/plugins/sdl/src/video/x11/SDL_x11dyn.c | 222 + apps/plugins/sdl/src/video/x11/SDL_x11dyn.h | 93 + apps/plugins/sdl/src/video/x11/SDL_x11events.c | 1414 ++++++ apps/plugins/sdl/src/video/x11/SDL_x11events_c.h | 34 + apps/plugins/sdl/src/video/x11/SDL_x11gamma.c | 142 + apps/plugins/sdl/src/video/x11/SDL_x11gamma_c.h | 32 + apps/plugins/sdl/src/video/x11/SDL_x11gl.c | 577 +++ apps/plugins/sdl/src/video/x11/SDL_x11gl_c.h | 99 + apps/plugins/sdl/src/video/x11/SDL_x11image.c | 316 ++ apps/plugins/sdl/src/video/x11/SDL_x11image_c.h | 38 + apps/plugins/sdl/src/video/x11/SDL_x11modes.c | 1143 +++++ apps/plugins/sdl/src/video/x11/SDL_x11modes_c.h | 43 + apps/plugins/sdl/src/video/x11/SDL_x11mouse.c | 288 ++ apps/plugins/sdl/src/video/x11/SDL_x11mouse_c.h | 33 + apps/plugins/sdl/src/video/x11/SDL_x11sym.h | 201 + apps/plugins/sdl/src/video/x11/SDL_x11video.c | 1571 ++++++ apps/plugins/sdl/src/video/x11/SDL_x11video.h | 214 + apps/plugins/sdl/src/video/x11/SDL_x11wm.c | 434 ++ apps/plugins/sdl/src/video/x11/SDL_x11wm_c.h | 34 + apps/plugins/sdl/src/video/x11/SDL_x11yuv.c | 538 +++ apps/plugins/sdl/src/video/x11/SDL_x11yuv_c.h | 41 + apps/plugins/sdl/src/video/xbios/SDL_xbios.c | 1116 +++++ apps/plugins/sdl/src/video/xbios/SDL_xbios.h | 111 + .../plugins/sdl/src/video/xbios/SDL_xbios_blowup.c | 77 + .../plugins/sdl/src/video/xbios/SDL_xbios_blowup.h | 86 + .../sdl/src/video/xbios/SDL_xbios_centscreen.c | 104 + .../sdl/src/video/xbios/SDL_xbios_centscreen.h | 114 + apps/plugins/sdl/src/video/xbios/SDL_xbios_milan.c | 106 + apps/plugins/sdl/src/video/xbios/SDL_xbios_milan.h | 129 + apps/plugins/sdl/src/video/xbios/SDL_xbios_sb3.c | 83 + apps/plugins/sdl/src/video/xbios/SDL_xbios_sb3.h | 82 + .../sdl/src/video/xbios/SDL_xbios_tveille.c | 63 + .../sdl/src/video/xbios/SDL_xbios_tveille.h | 64 + 643 files changed, 180278 insertions(+) create mode 100644 apps/plugins/sdl/src/SDL.c create mode 100644 apps/plugins/sdl/src/SDL_error.c create mode 100644 apps/plugins/sdl/src/SDL_error_c.h create mode 100644 apps/plugins/sdl/src/SDL_fatal.c create mode 100644 apps/plugins/sdl/src/SDL_fatal.h create mode 100644 apps/plugins/sdl/src/audio/SDL_audio.c create mode 100644 apps/plugins/sdl/src/audio/SDL_audio_c.h create mode 100644 apps/plugins/sdl/src/audio/SDL_audiocvt.c create mode 100644 apps/plugins/sdl/src/audio/SDL_audiodev.c create mode 100644 apps/plugins/sdl/src/audio/SDL_audiodev_c.h create mode 100644 apps/plugins/sdl/src/audio/SDL_audiomem.h create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer.c create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer_MMX.c create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer_MMX.h create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.c create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.h create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer_m68k.c create mode 100644 apps/plugins/sdl/src/audio/SDL_mixer_m68k.h create mode 100644 apps/plugins/sdl/src/audio/SDL_sysaudio.h create mode 100644 apps/plugins/sdl/src/audio/SDL_wave.c create mode 100644 apps/plugins/sdl/src/audio/SDL_wave.h create mode 100644 apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c create mode 100644 apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h create mode 100644 apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c create mode 100644 apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h create mode 100644 apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc create mode 100644 apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h create mode 100644 apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c create mode 100644 apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h create mode 100644 apps/plugins/sdl/src/audio/dart/SDL_dart.c create mode 100644 apps/plugins/sdl/src/audio/dart/SDL_dart.h create mode 100644 apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c create mode 100644 apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h create mode 100644 apps/plugins/sdl/src/audio/dc/aica.c create mode 100644 apps/plugins/sdl/src/audio/dc/aica.h create mode 100644 apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c create mode 100644 apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h create mode 100644 apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c create mode 100644 apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h create mode 100644 apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c create mode 100644 apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h create mode 100644 apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c create mode 100644 apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h create mode 100644 apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.c create mode 100644 apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.h create mode 100644 apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c create mode 100644 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apps/plugins/sdl/src/video/windx5/SDL_dx5yuv.c create mode 100644 apps/plugins/sdl/src/video/windx5/SDL_dx5yuv_c.h create mode 100644 apps/plugins/sdl/src/video/windx5/directx.h create mode 100644 apps/plugins/sdl/src/video/wscons/SDL_wsconsevents.c create mode 100644 apps/plugins/sdl/src/video/wscons/SDL_wsconsevents_c.h create mode 100644 apps/plugins/sdl/src/video/wscons/SDL_wsconsmouse.c create mode 100644 apps/plugins/sdl/src/video/wscons/SDL_wsconsmouse_c.h create mode 100644 apps/plugins/sdl/src/video/wscons/SDL_wsconsvideo.c create mode 100644 apps/plugins/sdl/src/video/wscons/SDL_wsconsvideo.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11dga.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11dga_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11dyn.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11dyn.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11events.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11events_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11gamma.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11gamma_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11gl.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11gl_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11image.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11image_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11modes.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11modes_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11mouse.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11mouse_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11sym.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11video.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11video.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11wm.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11wm_c.h create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11yuv.c create mode 100644 apps/plugins/sdl/src/video/x11/SDL_x11yuv_c.h create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios.c create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios.h create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_blowup.c create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_blowup.h create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_centscreen.c create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_centscreen.h create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_milan.c create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_milan.h create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_sb3.c create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_sb3.h create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_tveille.c create mode 100644 apps/plugins/sdl/src/video/xbios/SDL_xbios_tveille.h (limited to 'apps/plugins/sdl/src') diff --git a/apps/plugins/sdl/src/SDL.c b/apps/plugins/sdl/src/SDL.c new file mode 100644 index 0000000000..87f1b1af2a --- /dev/null +++ b/apps/plugins/sdl/src/SDL.c @@ -0,0 +1,350 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Initialization code for SDL */ + +#include "SDL.h" +#include "SDL_fatal.h" +#if !SDL_VIDEO_DISABLED +#include "video/SDL_leaks.h" +#endif + +#if SDL_THREAD_PTH +#include +#endif + +/* Initialization/Cleanup routines */ +#if !SDL_JOYSTICK_DISABLED +extern int SDL_JoystickInit(void); +extern void SDL_JoystickQuit(void); +#endif +#if !SDL_CDROM_DISABLED +extern int SDL_CDROMInit(void); +extern void SDL_CDROMQuit(void); +#endif +#if !SDL_TIMERS_DISABLED +extern void SDL_StartTicks(void); +extern int SDL_TimerInit(void); +extern void SDL_TimerQuit(void); +#endif + +/* The current SDL version */ +static SDL_version version = + { SDL_MAJOR_VERSION, SDL_MINOR_VERSION, SDL_PATCHLEVEL }; + +/* The initialized subsystems */ +static Uint32 SDL_initialized = 0; +#if !SDL_TIMERS_DISABLED +static Uint32 ticks_started = 0; +#endif + +#ifdef CHECK_LEAKS +int surfaces_allocated = 0; +#endif + +int SDL_InitSubSystem(Uint32 flags) +{ +#if !SDL_TIMERS_DISABLED + /* Initialize the timer subsystem */ + if ( ! ticks_started ) { + SDL_StartTicks(); + ticks_started = 1; + } + if ( (flags & SDL_INIT_TIMER) && !(SDL_initialized & SDL_INIT_TIMER) ) { + if ( SDL_TimerInit() < 0 ) { + return(-1); + } + SDL_initialized |= SDL_INIT_TIMER; + } +#else + if ( flags & SDL_INIT_TIMER ) { + SDL_SetError("SDL not built with timer support"); + return(-1); + } +#endif + +#if !SDL_VIDEO_DISABLED + /* Initialize the video/event subsystem */ + if ( (flags & SDL_INIT_VIDEO) && !(SDL_initialized & SDL_INIT_VIDEO) ) { + if ( SDL_VideoInit(SDL_getenv("SDL_VIDEODRIVER"), + (flags&SDL_INIT_EVENTTHREAD)) < 0 ) { + return(-1); + } + SDL_initialized |= SDL_INIT_VIDEO; + } +#else + if ( flags & SDL_INIT_VIDEO ) { + SDL_SetError("SDL not built with video support"); + return(-1); + } +#endif + +#if !SDL_AUDIO_DISABLED + /* Initialize the audio subsystem */ + if ( (flags & SDL_INIT_AUDIO) && !(SDL_initialized & SDL_INIT_AUDIO) ) { + if ( SDL_AudioInit(SDL_getenv("SDL_AUDIODRIVER")) < 0 ) { + return(-1); + } + SDL_initialized |= SDL_INIT_AUDIO; + } +#else + if ( flags & SDL_INIT_AUDIO ) { + SDL_SetError("SDL not built with audio support"); + return(-1); + } +#endif + +#if !SDL_JOYSTICK_DISABLED + /* Initialize the joystick subsystem */ + if ( (flags & SDL_INIT_JOYSTICK) && + !(SDL_initialized & SDL_INIT_JOYSTICK) ) { + if ( SDL_JoystickInit() < 0 ) { + return(-1); + } + SDL_initialized |= SDL_INIT_JOYSTICK; + } +#else + if ( flags & SDL_INIT_JOYSTICK ) { + SDL_SetError("SDL not built with joystick support"); + return(-1); + } +#endif + +#if !SDL_CDROM_DISABLED + /* Initialize the CD-ROM subsystem */ + if ( (flags & SDL_INIT_CDROM) && !(SDL_initialized & SDL_INIT_CDROM) ) { + if ( SDL_CDROMInit() < 0 ) { + return(-1); + } + SDL_initialized |= SDL_INIT_CDROM; + } +#else + if ( flags & SDL_INIT_CDROM ) { + SDL_SetError("SDL not built with cdrom support"); + return(-1); + } +#endif + return(0); +} + +int SDL_Init(Uint32 flags) +{ +#if !SDL_THREADS_DISABLED && SDL_THREAD_PTH + if (!pth_init()) { + return -1; + } +#endif + + /* Clear the error message */ + SDL_ClearError(); + + /* Initialize the desired subsystems */ + if ( SDL_InitSubSystem(flags) < 0 ) { + return(-1); + } + + /* Everything is initialized */ + if ( !(flags & SDL_INIT_NOPARACHUTE) ) { + SDL_InstallParachute(); + } + return(0); +} + +void SDL_QuitSubSystem(Uint32 flags) +{ + /* Shut down requested initialized subsystems */ +#if !SDL_CDROM_DISABLED + if ( (flags & SDL_initialized & SDL_INIT_CDROM) ) { + SDL_CDROMQuit(); + SDL_initialized &= ~SDL_INIT_CDROM; + } +#endif +#if !SDL_JOYSTICK_DISABLED + if ( (flags & SDL_initialized & SDL_INIT_JOYSTICK) ) { + SDL_JoystickQuit(); + SDL_initialized &= ~SDL_INIT_JOYSTICK; + } +#endif +#if !SDL_AUDIO_DISABLED + if ( (flags & SDL_initialized & SDL_INIT_AUDIO) ) { + SDL_AudioQuit(); + SDL_initialized &= ~SDL_INIT_AUDIO; + } +#endif +#if !SDL_VIDEO_DISABLED + if ( (flags & SDL_initialized & SDL_INIT_VIDEO) ) { + SDL_VideoQuit(); + SDL_initialized &= ~SDL_INIT_VIDEO; + } +#endif +#if !SDL_TIMERS_DISABLED + if ( (flags & SDL_initialized & SDL_INIT_TIMER) ) { + SDL_TimerQuit(); + SDL_initialized &= ~SDL_INIT_TIMER; + } +#endif +} + +Uint32 SDL_WasInit(Uint32 flags) +{ + if ( ! flags ) { + flags = SDL_INIT_EVERYTHING; + } + return (SDL_initialized&flags); +} + +void SDL_Quit(void) +{ + /* Quit all subsystems */ +#ifdef DEBUG_BUILD + printf("[SDL_Quit] : Enter! Calling QuitSubSystem()\n"); fflush(stdout); +#endif + SDL_QuitSubSystem(SDL_INIT_EVERYTHING); + +#ifdef CHECK_LEAKS +#ifdef DEBUG_BUILD + printf("[SDL_Quit] : CHECK_LEAKS\n"); fflush(stdout); +#endif + + /* Print the number of surfaces not freed */ + if ( surfaces_allocated != 0 ) { + fprintf(stderr, "SDL Warning: %d SDL surfaces extant\n", + surfaces_allocated); + } +#endif +#ifdef DEBUG_BUILD + printf("[SDL_Quit] : SDL_UninstallParachute()\n"); fflush(stdout); +#endif + + /* Uninstall any parachute signal handlers */ + SDL_UninstallParachute(); + +#if !SDL_THREADS_DISABLED && SDL_THREAD_PTH + pth_kill(); +#endif +#ifdef DEBUG_BUILD + printf("[SDL_Quit] : Returning!\n"); fflush(stdout); +#endif + +} + +/* Return the library version number */ +const SDL_version * SDL_Linked_Version(void) +{ + return(&version); +} + +#if defined(__OS2__) +/* Building for OS/2 */ +#ifdef __WATCOMC__ + +#define INCL_DOSERRORS +#define INCL_DOSEXCEPTIONS +#include + +/* Exception handler to prevent the Audio thread hanging, making a zombie process! */ +ULONG _System SDL_Main_ExceptionHandler(PEXCEPTIONREPORTRECORD pERepRec, + PEXCEPTIONREGISTRATIONRECORD pERegRec, + PCONTEXTRECORD pCtxRec, + PVOID p) +{ + if (pERepRec->fHandlerFlags & EH_EXIT_UNWIND) + return XCPT_CONTINUE_SEARCH; + if (pERepRec->fHandlerFlags & EH_UNWINDING) + return XCPT_CONTINUE_SEARCH; + if (pERepRec->fHandlerFlags & EH_NESTED_CALL) + return XCPT_CONTINUE_SEARCH; + + /* Do cleanup at every fatal exception! */ + if (((pERepRec->ExceptionNum & XCPT_SEVERITY_CODE) == XCPT_FATAL_EXCEPTION) && + (pERepRec->ExceptionNum != XCPT_BREAKPOINT) && + (pERepRec->ExceptionNum != XCPT_SINGLE_STEP) + ) + { + if (SDL_initialized & SDL_INIT_AUDIO) + { + /* This removes the zombie audio thread in case of emergency. */ +#ifdef DEBUG_BUILD + printf("[SDL_Main_ExceptionHandler] : Calling SDL_CloseAudio()!\n"); +#endif + SDL_CloseAudio(); + } + } + return (XCPT_CONTINUE_SEARCH); +} + + +EXCEPTIONREGISTRATIONRECORD SDL_Main_xcpthand = {0, SDL_Main_ExceptionHandler}; + +/* The main DLL entry for DLL Initialization and Uninitialization: */ +unsigned _System LibMain(unsigned hmod, unsigned termination) +{ + if (termination) + { +#ifdef DEBUG_BUILD +/* printf("[SDL DLL Unintialization] : Removing exception handler\n"); */ +#endif + DosUnsetExceptionHandler(&SDL_Main_xcpthand); + return 1; + } else + { +#ifdef DEBUG_BUILD + /* Make stdout and stderr unbuffered! */ + setbuf(stdout, NULL); + setbuf(stderr, NULL); +#endif + /* Fire up exception handler */ +#ifdef DEBUG_BUILD +/* printf("[SDL DLL Initialization] : Setting exception handler\n"); */ +#endif + /* Set exception handler */ + DosSetExceptionHandler(&SDL_Main_xcpthand); + + return 1; + } +} +#endif /* __WATCOMC__ */ + +#elif defined(__WIN32__) && !defined(__SYMBIAN32__) + +#if !defined(HAVE_LIBC) || (defined(__WATCOMC__) && defined(BUILD_DLL)) +/* Need to include DllMain() on Watcom C for some reason.. */ +#define WIN32_LEAN_AND_MEAN +#include + +BOOL APIENTRY _DllMainCRTStartup( HANDLE hModule, + DWORD ul_reason_for_call, + LPVOID lpReserved ) +{ + switch (ul_reason_for_call) { + case DLL_PROCESS_ATTACH: + case DLL_THREAD_ATTACH: + case DLL_THREAD_DETACH: + case DLL_PROCESS_DETACH: + break; + } + return TRUE; +} +#endif /* building DLL with Watcom C */ + +#endif /* OS/2 elif __WIN32__ */ diff --git a/apps/plugins/sdl/src/SDL_error.c b/apps/plugins/sdl/src/SDL_error.c new file mode 100644 index 0000000000..0f06bd53a3 --- /dev/null +++ b/apps/plugins/sdl/src/SDL_error.c @@ -0,0 +1,238 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Simple error handling in SDL */ + +#include "SDL_error.h" +#include "SDL_error_c.h" + +/* Routine to get the thread-specific error variable */ +#if SDL_THREADS_DISABLED +/* The SDL_arraysize(The ),default (non-thread-safe) global error variable */ +static SDL_error SDL_global_error; +#define SDL_GetErrBuf() (&SDL_global_error) +#else +extern SDL_error *SDL_GetErrBuf(void); +#endif /* SDL_THREADS_DISABLED */ + +#define SDL_ERRBUFIZE 1024 + +/* Private functions */ + +static const char *SDL_LookupString(const char *key) +{ + /* FIXME: Add code to lookup key in language string hash-table */ + return key; +} + +/* Public functions */ + +void SDL_SetError (const char *fmt, ...) +{ + va_list ap; + SDL_error *error; + + /* Copy in the key, mark error as valid */ + error = SDL_GetErrBuf(); + error->error = 1; + SDL_strlcpy((char *)error->key, fmt, sizeof(error->key)); + + va_start(ap, fmt); + error->argc = 0; + while ( *fmt ) { + if ( *fmt++ == '%' ) { + while ( *fmt == '.' || (*fmt >= '0' && *fmt <= '9') ) { + ++fmt; + } + switch (*fmt++) { + case 0: /* Malformed format string.. */ + --fmt; + break; + case 'c': + case 'i': + case 'd': + case 'u': + case 'o': + case 'x': + case 'X': + error->args[error->argc++].value_i = + va_arg(ap, int); + break; + case 'f': + error->args[error->argc++].value_f = + va_arg(ap, double); + break; + case 'p': + error->args[error->argc++].value_ptr = + va_arg(ap, void *); + break; + case 's': + { + int i = error->argc; + const char *str = va_arg(ap, const char *); + if (str == NULL) + str = "(null)"; + SDL_strlcpy((char *)error->args[i].buf, str, ERR_MAX_STRLEN); + error->argc++; + } + break; + default: + break; + } + if ( error->argc >= ERR_MAX_ARGS ) { + break; + } + } + } + va_end(ap); + + /* If we are in debug mode, print out an error message */ +#ifdef DEBUG_ERROR + fprintf(stderr, "SDL_SetError: %s\n", SDL_GetError()); +#endif +} + +/* This function has a bit more overhead than most error functions + so that it supports internationalization and thread-safe errors. +*/ +char *SDL_GetErrorMsg(char *errstr, unsigned int maxlen) +{ + SDL_error *error; + + /* Clear the error string */ + *errstr = '\0'; --maxlen; + + /* Get the thread-safe error, and print it out */ + error = SDL_GetErrBuf(); + if ( error->error ) { + const char *fmt; + char *msg = errstr; + int len; + int argi; + + fmt = SDL_LookupString(error->key); + argi = 0; + while ( *fmt && (maxlen > 0) ) { + if ( *fmt == '%' ) { + char tmp[32], *spot = tmp; + *spot++ = *fmt++; + while ( (*fmt == '.' || (*fmt >= '0' && *fmt <= '9')) && spot < (tmp+SDL_arraysize(tmp)-2) ) { + *spot++ = *fmt++; + } + *spot++ = *fmt++; + *spot++ = '\0'; + switch (spot[-2]) { + case '%': + *msg++ = '%'; + maxlen -= 1; + break; + case 'c': + case 'i': + case 'd': + case 'u': + case 'o': + case 'x': + case 'X': + len = SDL_snprintf(msg, maxlen, tmp, error->args[argi++].value_i); + msg += len; + maxlen -= len; + break; + case 'f': + len = SDL_snprintf(msg, maxlen, tmp, error->args[argi++].value_f); + msg += len; + maxlen -= len; + break; + case 'p': + len = SDL_snprintf(msg, maxlen, tmp, error->args[argi++].value_ptr); + msg += len; + maxlen -= len; + break; + case 's': + len = SDL_snprintf(msg, maxlen, tmp, SDL_LookupString(error->args[argi++].buf)); + msg += len; + maxlen -= len; + break; + } + } else { + *msg++ = *fmt++; + maxlen -= 1; + } + } + *msg = 0; /* NULL terminate the string */ + } + return(errstr); +} + +/* Available for backwards compatibility */ +char *SDL_GetError (void) +{ + static char errmsg[SDL_ERRBUFIZE]; + + return((char *)SDL_GetErrorMsg(errmsg, SDL_ERRBUFIZE)); +} + +void SDL_ClearError(void) +{ + SDL_error *error; + + error = SDL_GetErrBuf(); + error->error = 0; +} + +/* Very common errors go here */ +void SDL_Error(SDL_errorcode code) +{ + switch (code) { + case SDL_ENOMEM: + SDL_SetError("Out of memory"); + break; + case SDL_EFREAD: + SDL_SetError("Error reading from datastream"); + break; + case SDL_EFWRITE: + SDL_SetError("Error writing to datastream"); + break; + case SDL_EFSEEK: + SDL_SetError("Error seeking in datastream"); + break; + default: + SDL_SetError("Unknown SDL error"); + break; + } +} + +#ifdef TEST_ERROR +int main(int argc, char *argv[]) +{ + char buffer[BUFSIZ+1]; + + SDL_SetError("Hi there!"); + printf("Error 1: %s\n", SDL_GetError()); + SDL_ClearError(); + SDL_memset(buffer, '1', BUFSIZ); + buffer[BUFSIZ] = 0; + SDL_SetError("This is the error: %s (%f)", buffer, 1.0); + printf("Error 2: %s\n", SDL_GetError()); + exit(0); +} +#endif diff --git a/apps/plugins/sdl/src/SDL_error_c.h b/apps/plugins/sdl/src/SDL_error_c.h new file mode 100644 index 0000000000..8e54e42541 --- /dev/null +++ b/apps/plugins/sdl/src/SDL_error_c.h @@ -0,0 +1,58 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* This file defines a structure that carries language-independent + error messages +*/ + +#ifndef _SDL_error_c_h +#define _SDL_error_c_h + +#define ERR_MAX_STRLEN 128 +#define ERR_MAX_ARGS 5 + +typedef struct SDL_error { + /* This is a numeric value corresponding to the current error */ + int error; + + /* This is a key used to index into a language hashtable containing + internationalized versions of the SDL error messages. If the key + is not in the hashtable, or no hashtable is available, the key is + used directly as an error message format string. + */ + char key[ERR_MAX_STRLEN]; + + /* These are the arguments for the error functions */ + int argc; + union { + void *value_ptr; +#if 0 /* What is a character anyway? (UNICODE issues) */ + unsigned char value_c; +#endif + int value_i; + double value_f; + char buf[ERR_MAX_STRLEN]; + } args[ERR_MAX_ARGS]; +} SDL_error; + +#endif /* _SDL_error_c_h */ diff --git a/apps/plugins/sdl/src/SDL_fatal.c b/apps/plugins/sdl/src/SDL_fatal.c new file mode 100644 index 0000000000..d422a01f88 --- /dev/null +++ b/apps/plugins/sdl/src/SDL_fatal.c @@ -0,0 +1,134 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* General fatal signal handling code for SDL */ + +#ifdef HAVE_SIGNAL_H + +#include + +#include "SDL.h" +#include "SDL_fatal.h" + +/* This installs some signal handlers for the more common fatal signals, + so that if the programmer is lazy, the app doesn't die so horribly if + the program crashes. +*/ + +static void SDL_Parachute(int sig) +{ + signal(sig, SIG_DFL); + SDL_Quit(); + raise(sig); +} + +static int SDL_fatal_signals[] = { + SIGSEGV, +#ifdef SIGBUS + SIGBUS, +#endif +#ifdef SIGFPE + SIGFPE, +#endif +#ifdef SIGQUIT + SIGQUIT, +#endif + 0 +}; + +void SDL_InstallParachute(void) +{ + /* Set a handler for any fatal signal not already handled */ + int i; +#ifdef HAVE_SIGACTION + struct sigaction action; + + for ( i=0; SDL_fatal_signals[i]; ++i ) { + sigaction(SDL_fatal_signals[i], NULL, &action); + if ( action.sa_handler == SIG_DFL ) { + action.sa_handler = SDL_Parachute; + sigaction(SDL_fatal_signals[i], &action, NULL); + } + } +#ifdef SIGALRM + /* Set SIGALRM to be ignored -- necessary on Solaris */ + sigaction(SIGALRM, NULL, &action); + if ( action.sa_handler == SIG_DFL ) { + action.sa_handler = SIG_IGN; + sigaction(SIGALRM, &action, NULL); + } +#endif +#else + void (*ohandler)(int); + + for ( i=0; SDL_fatal_signals[i]; ++i ) { + ohandler = signal(SDL_fatal_signals[i], SDL_Parachute); + if ( ohandler != SIG_DFL ) { + signal(SDL_fatal_signals[i], ohandler); + } + } +#endif /* HAVE_SIGACTION */ + return; +} + +void SDL_UninstallParachute(void) +{ + /* Remove a handler for any fatal signal handled */ + int i; +#ifdef HAVE_SIGACTION + struct sigaction action; + + for ( i=0; SDL_fatal_signals[i]; ++i ) { + sigaction(SDL_fatal_signals[i], NULL, &action); + if ( action.sa_handler == SDL_Parachute ) { + action.sa_handler = SIG_DFL; + sigaction(SDL_fatal_signals[i], &action, NULL); + } + } +#else + void (*ohandler)(int); + + for ( i=0; SDL_fatal_signals[i]; ++i ) { + ohandler = signal(SDL_fatal_signals[i], SIG_DFL); + if ( ohandler != SDL_Parachute ) { + signal(SDL_fatal_signals[i], ohandler); + } + } +#endif /* HAVE_SIGACTION */ +} + +#else + +/* No signals on this platform, nothing to do.. */ + +void SDL_InstallParachute(void) +{ + return; +} + +void SDL_UninstallParachute(void) +{ + return; +} + +#endif /* HAVE_SIGNAL_H */ diff --git a/apps/plugins/sdl/src/SDL_fatal.h b/apps/plugins/sdl/src/SDL_fatal.h new file mode 100644 index 0000000000..6789784aeb --- /dev/null +++ b/apps/plugins/sdl/src/SDL_fatal.h @@ -0,0 +1,28 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* General fatal signal handling code for SDL */ + +extern void SDL_InstallParachute(void); +extern void SDL_UninstallParachute(void); + diff --git a/apps/plugins/sdl/src/audio/SDL_audio.c b/apps/plugins/sdl/src/audio/SDL_audio.c new file mode 100644 index 0000000000..9c0feeddfc --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_audio.c @@ -0,0 +1,707 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include "SDL.h" +#include "SDL_audio_c.h" +#include "SDL_audiomem.h" +#include "SDL_sysaudio.h" + +#ifdef __OS2__ +/* We'll need the DosSetPriority() API! */ +#define INCL_DOSPROCESS +#include +#endif + +/* Available audio drivers */ +static AudioBootStrap *bootstrap[] = { +#if SDL_AUDIO_DRIVER_PULSE + &PULSE_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_ALSA + &ALSA_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_BSD + &BSD_AUDIO_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_OSS + &DSP_bootstrap, + &DMA_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_QNXNTO + &QNXNTOAUDIO_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_SUNAUDIO + &SUNAUDIO_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_DMEDIA + &DMEDIA_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_ARTS + &ARTS_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_ESD + &ESD_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_NAS + &NAS_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_DSOUND + &DSOUND_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_WAVEOUT + &WAVEOUT_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_PAUD + &Paud_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_BAUDIO + &BAUDIO_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_COREAUDIO + &COREAUDIO_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_SNDMGR + &SNDMGR_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_MINT + &MINTAUDIO_GSXB_bootstrap, + &MINTAUDIO_MCSN_bootstrap, + &MINTAUDIO_STFA_bootstrap, + &MINTAUDIO_XBIOS_bootstrap, + &MINTAUDIO_DMA8_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_DISK + &DISKAUD_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_DUMMY + &DUMMYAUD_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_DC + &DCAUD_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_NDS + &NDSAUD_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_MMEAUDIO + &MMEAUDIO_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_DART + &DART_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_EPOCAUDIO + &EPOCAudio_bootstrap, +#endif +#if SDL_AUDIO_DRIVER_ROCKBOX + &ROCKBOXAUD_bootstrap, +#endif + NULL +}; +SDL_AudioDevice *current_audio = NULL; + +/* Various local functions */ +int SDL_AudioInit(const char *driver_name); +void SDL_AudioQuit(void); + +/* The general mixing thread function */ +int SDLCALL SDL_RunAudio(void *audiop) +{ + SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop; + Uint8 *stream; + int stream_len; + void *udata; + void (SDLCALL *fill)(void *userdata,Uint8 *stream, int len); + int silence; + + /* Perform any thread setup */ + if ( audio->ThreadInit ) { + audio->ThreadInit(audio); + } + audio->threadid = SDL_ThreadID(); + + /* Set up the mixing function */ + fill = audio->spec.callback; + udata = audio->spec.userdata; + + if ( audio->convert.needed ) { + if ( audio->convert.src_format == AUDIO_U8 ) { + silence = 0x80; + } else { + silence = 0; + } + stream_len = audio->convert.len; + } else { + silence = audio->spec.silence; + stream_len = audio->spec.size; + } + +#ifdef __OS2__ + /* Increase the priority of this thread to make sure that + the audio will be continuous all the time! */ +#ifdef USE_DOSSETPRIORITY + if (SDL_getenv("SDL_USE_TIMECRITICAL_AUDIO")) + { +#ifdef DEBUG_BUILD + printf("[SDL_RunAudio] : Setting priority to TimeCritical+0! (TID%d)\n", SDL_ThreadID()); +#endif + DosSetPriority(PRTYS_THREAD, PRTYC_TIMECRITICAL, 0, 0); + } + else + { +#ifdef DEBUG_BUILD + printf("[SDL_RunAudio] : Setting priority to ForegroundServer+0! (TID%d)\n", SDL_ThreadID()); +#endif + DosSetPriority(PRTYS_THREAD, PRTYC_FOREGROUNDSERVER, 0, 0); + } +#endif +#endif + + /* Loop, filling the audio buffers */ + while ( audio->enabled ) { + + /* Fill the current buffer with sound */ + if ( audio->convert.needed ) { + if ( audio->convert.buf ) { + stream = audio->convert.buf; + } else { + continue; + } + } else { + stream = audio->GetAudioBuf(audio); + if ( stream == NULL ) { + stream = audio->fake_stream; + } + } + + SDL_memset(stream, silence, stream_len); + + if ( ! audio->paused ) { + SDL_mutexP(audio->mixer_lock); + (*fill)(udata, stream, stream_len); + SDL_mutexV(audio->mixer_lock); + } + + /* Convert the audio if necessary */ + if ( audio->convert.needed ) { + SDL_ConvertAudio(&audio->convert); + stream = audio->GetAudioBuf(audio); + if ( stream == NULL ) { + stream = audio->fake_stream; + } + SDL_memcpy(stream, audio->convert.buf, + audio->convert.len_cvt); + } + + /* Ready current buffer for play and change current buffer */ + if ( stream != audio->fake_stream ) { + audio->PlayAudio(audio); + } + + /* Wait for an audio buffer to become available */ + if ( stream == audio->fake_stream ) { + SDL_Delay((audio->spec.samples*1000)/audio->spec.freq); + } else { + audio->WaitAudio(audio); + } + rb->yield(); + } + + /* Wait for the audio to drain.. */ + if ( audio->WaitDone ) { + audio->WaitDone(audio); + } + +#ifdef __OS2__ +#ifdef DEBUG_BUILD + printf("[SDL_RunAudio] : Task exiting. (TID%d)\n", SDL_ThreadID()); +#endif +#endif + return(0); +} + +static void SDL_LockAudio_Default(SDL_AudioDevice *audio) +{ + if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) { + return; + } + SDL_mutexP(audio->mixer_lock); +} + +static void SDL_UnlockAudio_Default(SDL_AudioDevice *audio) +{ + if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) { + return; + } + SDL_mutexV(audio->mixer_lock); +} + +static Uint16 SDL_ParseAudioFormat(const char *string) +{ + Uint16 format = 0; + + switch (*string) { + case 'U': + ++string; + format |= 0x0000; + break; + case 'S': + ++string; + format |= 0x8000; + break; + default: + return 0; + } + switch (SDL_atoi(string)) { + case 8: + string += 1; + format |= 8; + break; + case 16: + string += 2; + format |= 16; + if ( SDL_strcmp(string, "LSB") == 0 +#if SDL_BYTEORDER == SDL_LIL_ENDIAN + || SDL_strcmp(string, "SYS") == 0 +#endif + ) { + format |= 0x0000; + } + if ( SDL_strcmp(string, "MSB") == 0 +#if SDL_BYTEORDER == SDL_BIG_ENDIAN + || SDL_strcmp(string, "SYS") == 0 +#endif + ) { + format |= 0x1000; + } + break; + default: + return 0; + } + return format; +} + +int SDL_AudioInit(const char *driver_name) +{ + SDL_AudioDevice *audio; + int i = 0, idx; + + /* Check to make sure we don't overwrite 'current_audio' */ + if ( current_audio != NULL ) { + SDL_AudioQuit(); + } + + /* Select the proper audio driver */ + audio = NULL; + idx = 0; +#if SDL_AUDIO_DRIVER_ESD + if ( (driver_name == NULL) && (SDL_getenv("ESPEAKER") != NULL) ) { + /* Ahem, we know that if ESPEAKER is set, user probably wants + to use ESD, but don't start it if it's not already running. + This probably isn't the place to do this, but... Shh! :) + */ + for ( i=0; bootstrap[i]; ++i ) { + if ( SDL_strcasecmp(bootstrap[i]->name, "esd") == 0 ) { +#ifdef HAVE_PUTENV + const char *esd_no_spawn; + + /* Don't start ESD if it's not running */ + esd_no_spawn = getenv("ESD_NO_SPAWN"); + if ( esd_no_spawn == NULL ) { + putenv("ESD_NO_SPAWN=1"); + } +#endif + if ( bootstrap[i]->available() ) { + audio = bootstrap[i]->create(0); + break; + } +#ifdef HAVE_UNSETENV + if ( esd_no_spawn == NULL ) { + unsetenv("ESD_NO_SPAWN"); + } +#endif + } + } + } +#endif /* SDL_AUDIO_DRIVER_ESD */ + if ( audio == NULL ) { + if ( driver_name != NULL ) { +#if 0 /* This will be replaced with a better driver selection API */ + if ( SDL_strrchr(driver_name, ':') != NULL ) { + idx = atoi(SDL_strrchr(driver_name, ':')+1); + } +#endif + for ( i=0; bootstrap[i]; ++i ) { + if (SDL_strcasecmp(bootstrap[i]->name, driver_name) == 0) { + if ( bootstrap[i]->available() ) { + audio=bootstrap[i]->create(idx); + break; + } + } + } + } else { + for ( i=0; bootstrap[i]; ++i ) { + if ( bootstrap[i]->available() ) { + audio = bootstrap[i]->create(idx); + if ( audio != NULL ) { + break; + } + } + } + } + if ( audio == NULL ) { + SDL_SetError("No available audio device"); +#if 0 /* Don't fail SDL_Init() if audio isn't available. + SDL_OpenAudio() will handle it at that point. *sigh* + */ + return(-1); +#endif + } + } + current_audio = audio; + if ( current_audio ) { + current_audio->name = bootstrap[i]->name; + if ( !current_audio->LockAudio && !current_audio->UnlockAudio ) { + current_audio->LockAudio = SDL_LockAudio_Default; + current_audio->UnlockAudio = SDL_UnlockAudio_Default; + } + } + return(0); +} + +char *SDL_AudioDriverName(char *namebuf, int maxlen) +{ + if ( current_audio != NULL ) { + SDL_strlcpy(namebuf, current_audio->name, maxlen); + return(namebuf); + } + return(NULL); +} + +int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained) +{ + SDL_AudioDevice *audio; + const char *env; + + /* Start up the audio driver, if necessary */ + if ( ! current_audio ) { + if ( (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) || + (current_audio == NULL) ) { + return(-1); + } + } + audio = current_audio; + + if (audio->opened) { + SDL_SetError("Audio device is already opened"); + return(-1); + } + + /* Verify some parameters */ + if ( desired->freq == 0 ) { + env = SDL_getenv("SDL_AUDIO_FREQUENCY"); + if ( env ) { + desired->freq = SDL_atoi(env); + } + } + if ( desired->freq == 0 ) { + /* Pick some default audio frequency */ + desired->freq = 22050; + } + if ( desired->format == 0 ) { + env = SDL_getenv("SDL_AUDIO_FORMAT"); + if ( env ) { + desired->format = SDL_ParseAudioFormat(env); + } + } + if ( desired->format == 0 ) { + /* Pick some default audio format */ + desired->format = AUDIO_S16; + } + if ( desired->channels == 0 ) { + env = SDL_getenv("SDL_AUDIO_CHANNELS"); + if ( env ) { + desired->channels = (Uint8)SDL_atoi(env); + } + } + if ( desired->channels == 0 ) { + /* Pick a default number of channels */ + desired->channels = 2; + } + switch ( desired->channels ) { + case 1: /* Mono */ + case 2: /* Stereo */ + case 4: /* surround */ + case 6: /* surround with center and lfe */ + break; + default: + SDL_SetError("1 (mono) and 2 (stereo) channels supported"); + return(-1); + } + if ( desired->samples == 0 ) { + env = SDL_getenv("SDL_AUDIO_SAMPLES"); + if ( env ) { + desired->samples = (Uint16)SDL_atoi(env); + } + } + if ( desired->samples == 0 ) { + /* Pick a default of ~46 ms at desired frequency */ + int samples = (desired->freq / 1000) * 46; + int power2 = 1; + while ( power2 < samples ) { + power2 *= 2; + } + desired->samples = power2; + } + if ( desired->callback == NULL ) { + SDL_SetError("SDL_OpenAudio() passed a NULL callback"); + return(-1); + } + +#if SDL_THREADS_DISABLED + /* Uses interrupt driven audio, without thread */ +#else + /* Create a semaphore for locking the sound buffers */ + audio->mixer_lock = SDL_CreateMutex(); + if ( audio->mixer_lock == NULL ) { + SDL_SetError("Couldn't create mixer lock"); + SDL_CloseAudio(); + return(-1); + } +#endif /* SDL_THREADS_DISABLED */ + + /* Calculate the silence and size of the audio specification */ + SDL_CalculateAudioSpec(desired); + + /* Open the audio subsystem */ + SDL_memcpy(&audio->spec, desired, sizeof(audio->spec)); + audio->convert.needed = 0; + audio->enabled = 1; + audio->paused = 1; + + audio->opened = audio->OpenAudio(audio, &audio->spec)+1; + + if ( ! audio->opened ) { + SDL_CloseAudio(); + return(-1); + } + + /* If the audio driver changes the buffer size, accept it */ + if ( audio->spec.samples != desired->samples ) { + desired->samples = audio->spec.samples; + SDL_CalculateAudioSpec(desired); + } + + /* Allocate a fake audio memory buffer */ + audio->fake_stream = SDL_AllocAudioMem(audio->spec.size); + if ( audio->fake_stream == NULL ) { + SDL_CloseAudio(); + SDL_OutOfMemory(); + return(-1); + } + + /* See if we need to do any conversion */ + if ( obtained != NULL ) { + SDL_memcpy(obtained, &audio->spec, sizeof(audio->spec)); + } else if ( desired->freq != audio->spec.freq || + desired->format != audio->spec.format || + desired->channels != audio->spec.channels ) { + /* Build an audio conversion block */ + if ( SDL_BuildAudioCVT(&audio->convert, + desired->format, desired->channels, + desired->freq, + audio->spec.format, audio->spec.channels, + audio->spec.freq) < 0 ) { + SDL_CloseAudio(); + return(-1); + } + if ( audio->convert.needed ) { + audio->convert.len = (int) ( ((double) audio->spec.size) / + audio->convert.len_ratio ); + audio->convert.buf =(Uint8 *)SDL_AllocAudioMem( + audio->convert.len*audio->convert.len_mult); + if ( audio->convert.buf == NULL ) { + SDL_CloseAudio(); + SDL_OutOfMemory(); + return(-1); + } + } + } + + /* Start the audio thread if necessary */ + switch (audio->opened) { + case 1: + /* Start the audio thread */ +#if (defined(__WIN32__) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC) && !defined(__SYMBIAN32__) +#undef SDL_CreateThread + audio->thread = SDL_CreateThread(SDL_RunAudio, audio, NULL, NULL); +#else + audio->thread = SDL_CreateThread(SDL_RunAudio, audio); +#endif + if ( audio->thread == NULL ) { + SDL_CloseAudio(); + SDL_SetError("Couldn't create audio thread"); + return(-1); + } + break; + + default: + /* The audio is now playing */ + break; + } + + return(0); +} + +SDL_audiostatus SDL_GetAudioStatus(void) +{ + SDL_AudioDevice *audio = current_audio; + SDL_audiostatus status; + + status = SDL_AUDIO_STOPPED; + if ( audio && audio->enabled ) { + if ( audio->paused ) { + status = SDL_AUDIO_PAUSED; + } else { + status = SDL_AUDIO_PLAYING; + } + } + return(status); +} + +void SDL_PauseAudio (int pause_on) +{ + SDL_AudioDevice *audio = current_audio; + + if ( audio ) { + audio->paused = pause_on; + } +} + +void SDL_LockAudio (void) +{ + SDL_AudioDevice *audio = current_audio; + + /* Obtain a lock on the mixing buffers */ + if ( audio && audio->LockAudio ) { + audio->LockAudio(audio); + } +} + +void SDL_UnlockAudio (void) +{ + SDL_AudioDevice *audio = current_audio; + + /* Release lock on the mixing buffers */ + if ( audio && audio->UnlockAudio ) { + audio->UnlockAudio(audio); + } +} + +void SDL_CloseAudio (void) +{ + SDL_QuitSubSystem(SDL_INIT_AUDIO); +} + +void SDL_AudioQuit(void) +{ + SDL_AudioDevice *audio = current_audio; + + if ( audio ) { + audio->enabled = 0; + if ( audio->thread != NULL ) { + SDL_WaitThread(audio->thread, NULL); + } + if ( audio->mixer_lock != NULL ) { + SDL_DestroyMutex(audio->mixer_lock); + } + if ( audio->fake_stream != NULL ) { + SDL_FreeAudioMem(audio->fake_stream); + } + if ( audio->convert.needed ) { + SDL_FreeAudioMem(audio->convert.buf); + + } + if ( audio->opened ) { + audio->CloseAudio(audio); + audio->opened = 0; + } + /* Free the driver data */ + audio->free(audio); + current_audio = NULL; + } +} + +#define NUM_FORMATS 6 +static int format_idx; +static int format_idx_sub; +static Uint16 format_list[NUM_FORMATS][NUM_FORMATS] = { + { AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB }, + { AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB }, + { AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 }, + { AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 }, + { AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U8, AUDIO_S8 }, + { AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U8, AUDIO_S8 }, +}; + +Uint16 SDL_FirstAudioFormat(Uint16 format) +{ + for ( format_idx=0; format_idx < NUM_FORMATS; ++format_idx ) { + if ( format_list[format_idx][0] == format ) { + break; + } + } + format_idx_sub = 0; + return(SDL_NextAudioFormat()); +} + +Uint16 SDL_NextAudioFormat(void) +{ + if ( (format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS) ) { + return(0); + } + return(format_list[format_idx][format_idx_sub++]); +} + +void SDL_CalculateAudioSpec(SDL_AudioSpec *spec) +{ + switch (spec->format) { + case AUDIO_U8: + spec->silence = 0x80; + break; + default: + spec->silence = 0x00; + break; + } + spec->size = (spec->format&0xFF)/8; + spec->size *= spec->channels; + spec->size *= spec->samples; +} + +void SDL_Audio_SetCaption(const char *caption) +{ + if ((current_audio) && (current_audio->SetCaption)) { + current_audio->SetCaption(current_audio, caption); + } +} + diff --git a/apps/plugins/sdl/src/audio/SDL_audio_c.h b/apps/plugins/sdl/src/audio/SDL_audio_c.h new file mode 100644 index 0000000000..5fcf2029f7 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_audio_c.h @@ -0,0 +1,35 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */ + +/* Functions to get a list of "close" audio formats */ +extern Uint16 SDL_FirstAudioFormat(Uint16 format); +extern Uint16 SDL_NextAudioFormat(void); + +/* Function to calculate the size and silence for a SDL_AudioSpec */ +extern void SDL_CalculateAudioSpec(SDL_AudioSpec *spec); + +/* The actual mixing thread function */ +extern int SDLCALL SDL_RunAudio(void *audiop); + diff --git a/apps/plugins/sdl/src/audio/SDL_audiocvt.c b/apps/plugins/sdl/src/audio/SDL_audiocvt.c new file mode 100644 index 0000000000..9b8fbcd6e9 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_audiocvt.c @@ -0,0 +1,1510 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Functions for audio drivers to perform runtime conversion of audio format */ + +#include "SDL_audio.h" + + +/* Effectively mix right and left channels into a single channel */ +void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Sint32 sample; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting to mono\n"); +#endif + switch (format&0x8018) { + + case AUDIO_U8: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + for ( i=cvt->len_cvt/2; i; --i ) { + sample = src[0] + src[1]; + *dst = (Uint8)(sample / 2); + src += 2; + dst += 1; + } + } + break; + + case AUDIO_S8: { + Sint8 *src, *dst; + + src = (Sint8 *)cvt->buf; + dst = (Sint8 *)cvt->buf; + for ( i=cvt->len_cvt/2; i; --i ) { + sample = src[0] + src[1]; + *dst = (Sint8)(sample / 2); + src += 2; + dst += 1; + } + } + break; + + case AUDIO_U16: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/4; i; --i ) { + sample = (Uint16)((src[0]<<8)|src[1])+ + (Uint16)((src[2]<<8)|src[3]); + sample /= 2; + dst[1] = (sample&0xFF); + sample >>= 8; + dst[0] = (sample&0xFF); + src += 4; + dst += 2; + } + } else { + for ( i=cvt->len_cvt/4; i; --i ) { + sample = (Uint16)((src[1]<<8)|src[0])+ + (Uint16)((src[3]<<8)|src[2]); + sample /= 2; + dst[0] = (sample&0xFF); + sample >>= 8; + dst[1] = (sample&0xFF); + src += 4; + dst += 2; + } + } + } + break; + + case AUDIO_S16: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/4; i; --i ) { + sample = (Sint16)((src[0]<<8)|src[1])+ + (Sint16)((src[2]<<8)|src[3]); + sample /= 2; + dst[1] = (sample&0xFF); + sample >>= 8; + dst[0] = (sample&0xFF); + src += 4; + dst += 2; + } + } else { + for ( i=cvt->len_cvt/4; i; --i ) { + sample = (Sint16)((src[1]<<8)|src[0])+ + (Sint16)((src[3]<<8)|src[2]); + sample /= 2; + dst[0] = (sample&0xFF); + sample >>= 8; + dst[1] = (sample&0xFF); + src += 4; + dst += 2; + } + } + } + break; + } + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Discard top 4 channels */ +void SDLCALL SDL_ConvertStrip(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Sint32 lsample, rsample; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting down to stereo\n"); +#endif + switch (format&0x8018) { + + case AUDIO_U8: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + for ( i=cvt->len_cvt/6; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + src += 6; + dst += 2; + } + } + break; + + case AUDIO_S8: { + Sint8 *src, *dst; + + src = (Sint8 *)cvt->buf; + dst = (Sint8 *)cvt->buf; + for ( i=cvt->len_cvt/6; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + src += 6; + dst += 2; + } + } + break; + + case AUDIO_U16: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/12; i; --i ) { + lsample = (Uint16)((src[0]<<8)|src[1]); + rsample = (Uint16)((src[2]<<8)|src[3]); + dst[1] = (lsample&0xFF); + lsample >>= 8; + dst[0] = (lsample&0xFF); + dst[3] = (rsample&0xFF); + rsample >>= 8; + dst[2] = (rsample&0xFF); + src += 12; + dst += 4; + } + } else { + for ( i=cvt->len_cvt/12; i; --i ) { + lsample = (Uint16)((src[1]<<8)|src[0]); + rsample = (Uint16)((src[3]<<8)|src[2]); + dst[0] = (lsample&0xFF); + lsample >>= 8; + dst[1] = (lsample&0xFF); + dst[2] = (rsample&0xFF); + rsample >>= 8; + dst[3] = (rsample&0xFF); + src += 12; + dst += 4; + } + } + } + break; + + case AUDIO_S16: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/12; i; --i ) { + lsample = (Sint16)((src[0]<<8)|src[1]); + rsample = (Sint16)((src[2]<<8)|src[3]); + dst[1] = (lsample&0xFF); + lsample >>= 8; + dst[0] = (lsample&0xFF); + dst[3] = (rsample&0xFF); + rsample >>= 8; + dst[2] = (rsample&0xFF); + src += 12; + dst += 4; + } + } else { + for ( i=cvt->len_cvt/12; i; --i ) { + lsample = (Sint16)((src[1]<<8)|src[0]); + rsample = (Sint16)((src[3]<<8)|src[2]); + dst[0] = (lsample&0xFF); + lsample >>= 8; + dst[1] = (lsample&0xFF); + dst[2] = (rsample&0xFF); + rsample >>= 8; + dst[3] = (rsample&0xFF); + src += 12; + dst += 4; + } + } + } + break; + } + cvt->len_cvt /= 3; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Discard top 2 channels of 6 */ +void SDLCALL SDL_ConvertStrip_2(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Sint32 lsample, rsample; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting 6 down to quad\n"); +#endif + switch (format&0x8018) { + + case AUDIO_U8: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + for ( i=cvt->len_cvt/4; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + src += 4; + dst += 2; + } + } + break; + + case AUDIO_S8: { + Sint8 *src, *dst; + + src = (Sint8 *)cvt->buf; + dst = (Sint8 *)cvt->buf; + for ( i=cvt->len_cvt/4; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + src += 4; + dst += 2; + } + } + break; + + case AUDIO_U16: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/8; i; --i ) { + lsample = (Uint16)((src[0]<<8)|src[1]); + rsample = (Uint16)((src[2]<<8)|src[3]); + dst[1] = (lsample&0xFF); + lsample >>= 8; + dst[0] = (lsample&0xFF); + dst[3] = (rsample&0xFF); + rsample >>= 8; + dst[2] = (rsample&0xFF); + src += 8; + dst += 4; + } + } else { + for ( i=cvt->len_cvt/8; i; --i ) { + lsample = (Uint16)((src[1]<<8)|src[0]); + rsample = (Uint16)((src[3]<<8)|src[2]); + dst[0] = (lsample&0xFF); + lsample >>= 8; + dst[1] = (lsample&0xFF); + dst[2] = (rsample&0xFF); + rsample >>= 8; + dst[3] = (rsample&0xFF); + src += 8; + dst += 4; + } + } + } + break; + + case AUDIO_S16: { + Uint8 *src, *dst; + + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/8; i; --i ) { + lsample = (Sint16)((src[0]<<8)|src[1]); + rsample = (Sint16)((src[2]<<8)|src[3]); + dst[1] = (lsample&0xFF); + lsample >>= 8; + dst[0] = (lsample&0xFF); + dst[3] = (rsample&0xFF); + rsample >>= 8; + dst[2] = (rsample&0xFF); + src += 8; + dst += 4; + } + } else { + for ( i=cvt->len_cvt/8; i; --i ) { + lsample = (Sint16)((src[1]<<8)|src[0]); + rsample = (Sint16)((src[3]<<8)|src[2]); + dst[0] = (lsample&0xFF); + lsample >>= 8; + dst[1] = (lsample&0xFF); + dst[2] = (rsample&0xFF); + rsample >>= 8; + dst[3] = (rsample&0xFF); + src += 8; + dst += 4; + } + } + } + break; + } + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Duplicate a mono channel to both stereo channels */ +void SDLCALL SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting to stereo\n"); +#endif + if ( (format & 0xFF) == 16 ) { + Uint16 *src, *dst; + + src = (Uint16 *)(cvt->buf+cvt->len_cvt); + dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2); + for ( i=cvt->len_cvt/2; i; --i ) { + dst -= 2; + src -= 1; + dst[0] = src[0]; + dst[1] = src[0]; + } + } else { + Uint8 *src, *dst; + + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + for ( i=cvt->len_cvt; i; --i ) { + dst -= 2; + src -= 1; + dst[0] = src[0]; + dst[1] = src[0]; + } + } + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Duplicate a stereo channel to a pseudo-5.1 stream */ +void SDLCALL SDL_ConvertSurround(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting stereo to surround\n"); +#endif + switch (format&0x8018) { + + case AUDIO_U8: { + Uint8 *src, *dst, lf, rf, ce; + + src = (Uint8 *)(cvt->buf+cvt->len_cvt); + dst = (Uint8 *)(cvt->buf+cvt->len_cvt*3); + for ( i=cvt->len_cvt; i; --i ) { + dst -= 6; + src -= 2; + lf = src[0]; + rf = src[1]; + ce = (lf/2) + (rf/2); + dst[0] = lf; + dst[1] = rf; + dst[2] = lf - ce; + dst[3] = rf - ce; + dst[4] = ce; + dst[5] = ce; + } + } + break; + + case AUDIO_S8: { + Sint8 *src, *dst, lf, rf, ce; + + src = (Sint8 *)cvt->buf+cvt->len_cvt; + dst = (Sint8 *)cvt->buf+cvt->len_cvt*3; + for ( i=cvt->len_cvt; i; --i ) { + dst -= 6; + src -= 2; + lf = src[0]; + rf = src[1]; + ce = (lf/2) + (rf/2); + dst[0] = lf; + dst[1] = rf; + dst[2] = lf - ce; + dst[3] = rf - ce; + dst[4] = ce; + dst[5] = ce; + } + } + break; + + case AUDIO_U16: { + Uint8 *src, *dst; + Uint16 lf, rf, ce, lr, rr; + + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*3; + + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 12; + src -= 4; + lf = (Uint16)((src[0]<<8)|src[1]); + rf = (Uint16)((src[2]<<8)|src[3]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[1] = (lf&0xFF); + dst[0] = ((lf>>8)&0xFF); + dst[3] = (rf&0xFF); + dst[2] = ((rf>>8)&0xFF); + + dst[1+4] = (lr&0xFF); + dst[0+4] = ((lr>>8)&0xFF); + dst[3+4] = (rr&0xFF); + dst[2+4] = ((rr>>8)&0xFF); + + dst[1+8] = (ce&0xFF); + dst[0+8] = ((ce>>8)&0xFF); + dst[3+8] = (ce&0xFF); + dst[2+8] = ((ce>>8)&0xFF); + } + } else { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 12; + src -= 4; + lf = (Uint16)((src[1]<<8)|src[0]); + rf = (Uint16)((src[3]<<8)|src[2]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[0] = (lf&0xFF); + dst[1] = ((lf>>8)&0xFF); + dst[2] = (rf&0xFF); + dst[3] = ((rf>>8)&0xFF); + + dst[0+4] = (lr&0xFF); + dst[1+4] = ((lr>>8)&0xFF); + dst[2+4] = (rr&0xFF); + dst[3+4] = ((rr>>8)&0xFF); + + dst[0+8] = (ce&0xFF); + dst[1+8] = ((ce>>8)&0xFF); + dst[2+8] = (ce&0xFF); + dst[3+8] = ((ce>>8)&0xFF); + } + } + } + break; + + case AUDIO_S16: { + Uint8 *src, *dst; + Sint16 lf, rf, ce, lr, rr; + + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*3; + + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 12; + src -= 4; + lf = (Sint16)((src[0]<<8)|src[1]); + rf = (Sint16)((src[2]<<8)|src[3]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[1] = (lf&0xFF); + dst[0] = ((lf>>8)&0xFF); + dst[3] = (rf&0xFF); + dst[2] = ((rf>>8)&0xFF); + + dst[1+4] = (lr&0xFF); + dst[0+4] = ((lr>>8)&0xFF); + dst[3+4] = (rr&0xFF); + dst[2+4] = ((rr>>8)&0xFF); + + dst[1+8] = (ce&0xFF); + dst[0+8] = ((ce>>8)&0xFF); + dst[3+8] = (ce&0xFF); + dst[2+8] = ((ce>>8)&0xFF); + } + } else { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 12; + src -= 4; + lf = (Sint16)((src[1]<<8)|src[0]); + rf = (Sint16)((src[3]<<8)|src[2]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[0] = (lf&0xFF); + dst[1] = ((lf>>8)&0xFF); + dst[2] = (rf&0xFF); + dst[3] = ((rf>>8)&0xFF); + + dst[0+4] = (lr&0xFF); + dst[1+4] = ((lr>>8)&0xFF); + dst[2+4] = (rr&0xFF); + dst[3+4] = ((rr>>8)&0xFF); + + dst[0+8] = (ce&0xFF); + dst[1+8] = ((ce>>8)&0xFF); + dst[2+8] = (ce&0xFF); + dst[3+8] = ((ce>>8)&0xFF); + } + } + } + break; + } + cvt->len_cvt *= 3; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Duplicate a stereo channel to a pseudo-4.0 stream */ +void SDLCALL SDL_ConvertSurround_4(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting stereo to quad\n"); +#endif + switch (format&0x8018) { + + case AUDIO_U8: { + Uint8 *src, *dst, lf, rf, ce; + + src = (Uint8 *)(cvt->buf+cvt->len_cvt); + dst = (Uint8 *)(cvt->buf+cvt->len_cvt*2); + for ( i=cvt->len_cvt; i; --i ) { + dst -= 4; + src -= 2; + lf = src[0]; + rf = src[1]; + ce = (lf/2) + (rf/2); + dst[0] = lf; + dst[1] = rf; + dst[2] = lf - ce; + dst[3] = rf - ce; + } + } + break; + + case AUDIO_S8: { + Sint8 *src, *dst, lf, rf, ce; + + src = (Sint8 *)cvt->buf+cvt->len_cvt; + dst = (Sint8 *)cvt->buf+cvt->len_cvt*2; + for ( i=cvt->len_cvt; i; --i ) { + dst -= 4; + src -= 2; + lf = src[0]; + rf = src[1]; + ce = (lf/2) + (rf/2); + dst[0] = lf; + dst[1] = rf; + dst[2] = lf - ce; + dst[3] = rf - ce; + } + } + break; + + case AUDIO_U16: { + Uint8 *src, *dst; + Uint16 lf, rf, ce, lr, rr; + + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 8; + src -= 4; + lf = (Uint16)((src[0]<<8)|src[1]); + rf = (Uint16)((src[2]<<8)|src[3]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[1] = (lf&0xFF); + dst[0] = ((lf>>8)&0xFF); + dst[3] = (rf&0xFF); + dst[2] = ((rf>>8)&0xFF); + + dst[1+4] = (lr&0xFF); + dst[0+4] = ((lr>>8)&0xFF); + dst[3+4] = (rr&0xFF); + dst[2+4] = ((rr>>8)&0xFF); + } + } else { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 8; + src -= 4; + lf = (Uint16)((src[1]<<8)|src[0]); + rf = (Uint16)((src[3]<<8)|src[2]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[0] = (lf&0xFF); + dst[1] = ((lf>>8)&0xFF); + dst[2] = (rf&0xFF); + dst[3] = ((rf>>8)&0xFF); + + dst[0+4] = (lr&0xFF); + dst[1+4] = ((lr>>8)&0xFF); + dst[2+4] = (rr&0xFF); + dst[3+4] = ((rr>>8)&0xFF); + } + } + } + break; + + case AUDIO_S16: { + Uint8 *src, *dst; + Sint16 lf, rf, ce, lr, rr; + + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + + if ( (format & 0x1000) == 0x1000 ) { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 8; + src -= 4; + lf = (Sint16)((src[0]<<8)|src[1]); + rf = (Sint16)((src[2]<<8)|src[3]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[1] = (lf&0xFF); + dst[0] = ((lf>>8)&0xFF); + dst[3] = (rf&0xFF); + dst[2] = ((rf>>8)&0xFF); + + dst[1+4] = (lr&0xFF); + dst[0+4] = ((lr>>8)&0xFF); + dst[3+4] = (rr&0xFF); + dst[2+4] = ((rr>>8)&0xFF); + } + } else { + for ( i=cvt->len_cvt/4; i; --i ) { + dst -= 8; + src -= 4; + lf = (Sint16)((src[1]<<8)|src[0]); + rf = (Sint16)((src[3]<<8)|src[2]); + ce = (lf/2) + (rf/2); + rr = lf - ce; + lr = rf - ce; + dst[0] = (lf&0xFF); + dst[1] = ((lf>>8)&0xFF); + dst[2] = (rf&0xFF); + dst[3] = ((rf>>8)&0xFF); + + dst[0+4] = (lr&0xFF); + dst[1+4] = ((lr>>8)&0xFF); + dst[2+4] = (rr&0xFF); + dst[3+4] = ((rr>>8)&0xFF); + } + } + } + break; + } + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Convert 8-bit to 16-bit - LSB */ +void SDLCALL SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting to 16-bit LSB\n"); +#endif + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + for ( i=cvt->len_cvt; i; --i ) { + src -= 1; + dst -= 2; + dst[1] = *src; + dst[0] = 0; + } + format = ((format & ~0x0008) | AUDIO_U16LSB); + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} +/* Convert 8-bit to 16-bit - MSB */ +void SDLCALL SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting to 16-bit MSB\n"); +#endif + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + for ( i=cvt->len_cvt; i; --i ) { + src -= 1; + dst -= 2; + dst[0] = *src; + dst[1] = 0; + } + format = ((format & ~0x0008) | AUDIO_U16MSB); + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Convert 16-bit to 8-bit */ +void SDLCALL SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting to 8-bit\n"); +#endif + src = cvt->buf; + dst = cvt->buf; + if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ + ++src; + } + for ( i=cvt->len_cvt/2; i; --i ) { + *dst = *src; + src += 2; + dst += 1; + } + format = ((format & ~0x9010) | AUDIO_U8); + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Toggle signed/unsigned */ +void SDLCALL SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *data; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio signedness\n"); +#endif + data = cvt->buf; + if ( (format & 0xFF) == 16 ) { + if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ + ++data; + } + for ( i=cvt->len_cvt/2; i; --i ) { + *data ^= 0x80; + data += 2; + } + } else { + for ( i=cvt->len_cvt; i; --i ) { + *data++ ^= 0x80; + } + } + format = (format ^ 0x8000); + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Toggle endianness */ +void SDLCALL SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *data, tmp; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio endianness\n"); +#endif + data = cvt->buf; + for ( i=cvt->len_cvt/2; i; --i ) { + tmp = data[0]; + data[0] = data[1]; + data[1] = tmp; + data += 2; + } + format = (format ^ 0x1000); + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Convert rate up by multiple of 2 */ +void SDLCALL SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * 2\n"); +#endif + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt; i; --i ) { + src -= 1; + dst -= 2; + dst[0] = src[0]; + dst[1] = src[0]; + } + break; + case 16: + for ( i=cvt->len_cvt/2; i; --i ) { + src -= 2; + dst -= 4; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[0]; + dst[3] = src[1]; + } + break; + } + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Convert rate up by multiple of 2, for stereo */ +void SDLCALL SDL_RateMUL2_c2(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * 2\n"); +#endif + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/2; i; --i ) { + src -= 2; + dst -= 4; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[0]; + dst[3] = src[1]; + } + break; + case 16: + for ( i=cvt->len_cvt/4; i; --i ) { + src -= 4; + dst -= 8; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[0]; + dst[5] = src[1]; + dst[6] = src[2]; + dst[7] = src[3]; + } + break; + } + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Convert rate up by multiple of 2, for quad */ +void SDLCALL SDL_RateMUL2_c4(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * 2\n"); +#endif + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/4; i; --i ) { + src -= 4; + dst -= 8; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[0]; + dst[5] = src[1]; + dst[6] = src[2]; + dst[7] = src[3]; + } + break; + case 16: + for ( i=cvt->len_cvt/8; i; --i ) { + src -= 8; + dst -= 16; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[4]; + dst[5] = src[5]; + dst[6] = src[6]; + dst[7] = src[7]; + dst[8] = src[0]; + dst[9] = src[1]; + dst[10] = src[2]; + dst[11] = src[3]; + dst[12] = src[4]; + dst[13] = src[5]; + dst[14] = src[6]; + dst[15] = src[7]; + } + break; + } + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Convert rate up by multiple of 2, for 5.1 */ +void SDLCALL SDL_RateMUL2_c6(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * 2\n"); +#endif + src = cvt->buf+cvt->len_cvt; + dst = cvt->buf+cvt->len_cvt*2; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/6; i; --i ) { + src -= 6; + dst -= 12; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[4]; + dst[5] = src[5]; + dst[6] = src[0]; + dst[7] = src[1]; + dst[8] = src[2]; + dst[9] = src[3]; + dst[10] = src[4]; + dst[11] = src[5]; + } + break; + case 16: + for ( i=cvt->len_cvt/12; i; --i ) { + src -= 12; + dst -= 24; + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[4]; + dst[5] = src[5]; + dst[6] = src[6]; + dst[7] = src[7]; + dst[8] = src[8]; + dst[9] = src[9]; + dst[10] = src[10]; + dst[11] = src[11]; + dst[12] = src[0]; + dst[13] = src[1]; + dst[14] = src[2]; + dst[15] = src[3]; + dst[16] = src[4]; + dst[17] = src[5]; + dst[18] = src[6]; + dst[19] = src[7]; + dst[20] = src[8]; + dst[21] = src[9]; + dst[22] = src[10]; + dst[23] = src[11]; + } + break; + } + cvt->len_cvt *= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Convert rate down by multiple of 2 */ +void SDLCALL SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate / 2\n"); +#endif + src = cvt->buf; + dst = cvt->buf; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/2; i; --i ) { + dst[0] = src[0]; + src += 2; + dst += 1; + } + break; + case 16: + for ( i=cvt->len_cvt/4; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + src += 4; + dst += 2; + } + break; + } + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Convert rate down by multiple of 2, for stereo */ +void SDLCALL SDL_RateDIV2_c2(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate / 2\n"); +#endif + src = cvt->buf; + dst = cvt->buf; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/4; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + src += 4; + dst += 2; + } + break; + case 16: + for ( i=cvt->len_cvt/8; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + src += 8; + dst += 4; + } + break; + } + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + + +/* Convert rate down by multiple of 2, for quad */ +void SDLCALL SDL_RateDIV2_c4(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate / 2\n"); +#endif + src = cvt->buf; + dst = cvt->buf; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/8; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + src += 8; + dst += 4; + } + break; + case 16: + for ( i=cvt->len_cvt/16; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[4]; + dst[5] = src[5]; + dst[6] = src[6]; + dst[7] = src[7]; + src += 16; + dst += 8; + } + break; + } + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Convert rate down by multiple of 2, for 5.1 */ +void SDLCALL SDL_RateDIV2_c6(SDL_AudioCVT *cvt, Uint16 format) +{ + int i; + Uint8 *src, *dst; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate / 2\n"); +#endif + src = cvt->buf; + dst = cvt->buf; + switch (format & 0xFF) { + case 8: + for ( i=cvt->len_cvt/12; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[4]; + dst[5] = src[5]; + src += 12; + dst += 6; + } + break; + case 16: + for ( i=cvt->len_cvt/24; i; --i ) { + dst[0] = src[0]; + dst[1] = src[1]; + dst[2] = src[2]; + dst[3] = src[3]; + dst[4] = src[4]; + dst[5] = src[5]; + dst[6] = src[6]; + dst[7] = src[7]; + dst[8] = src[8]; + dst[9] = src[9]; + dst[10] = src[10]; + dst[11] = src[11]; + src += 24; + dst += 12; + } + break; + } + cvt->len_cvt /= 2; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +/* Very slow rate conversion routine */ +void SDLCALL SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format) +{ + double ipos; + int i, clen; + +#ifdef DEBUG_CONVERT + fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr); +#endif + clen = (int)((double)cvt->len_cvt / cvt->rate_incr); + if ( cvt->rate_incr > 1.0 ) { + switch (format & 0xFF) { + case 8: { + Uint8 *output; + + output = cvt->buf; + ipos = 0.0; + for ( i=clen; i; --i ) { + *output = cvt->buf[(int)ipos]; + ipos += cvt->rate_incr; + output += 1; + } + } + break; + + case 16: { + Uint16 *output; + + clen &= ~1; + output = (Uint16 *)cvt->buf; + ipos = 0.0; + for ( i=clen/2; i; --i ) { + *output=((Uint16 *)cvt->buf)[(int)ipos]; + ipos += cvt->rate_incr; + output += 1; + } + } + break; + } + } else { + switch (format & 0xFF) { + case 8: { + Uint8 *output; + + output = cvt->buf+clen; + ipos = (double)cvt->len_cvt; + for ( i=clen; i; --i ) { + ipos -= cvt->rate_incr; + output -= 1; + *output = cvt->buf[(int)ipos]; + } + } + break; + + case 16: { + Uint16 *output; + + clen &= ~1; + output = (Uint16 *)(cvt->buf+clen); + ipos = (double)cvt->len_cvt/2; + for ( i=clen/2; i; --i ) { + ipos -= cvt->rate_incr; + output -= 1; + *output=((Uint16 *)cvt->buf)[(int)ipos]; + } + } + break; + } + } + cvt->len_cvt = clen; + if ( cvt->filters[++cvt->filter_index] ) { + cvt->filters[cvt->filter_index](cvt, format); + } +} + +int SDL_ConvertAudio(SDL_AudioCVT *cvt) +{ + /* Make sure there's data to convert */ + if ( cvt->buf == NULL ) { + SDL_SetError("No buffer allocated for conversion"); + return(-1); + } + /* Return okay if no conversion is necessary */ + cvt->len_cvt = cvt->len; + if ( cvt->filters[0] == NULL ) { + return(0); + } + + /* Set up the conversion and go! */ + cvt->filter_index = 0; + cvt->filters[0](cvt, cvt->src_format); + return(0); +} + +/* Creates a set of audio filters to convert from one format to another. + Returns -1 if the format conversion is not supported, or 1 if the + audio filter is set up. +*/ + +int SDL_BuildAudioCVT(SDL_AudioCVT *cvt, + Uint16 src_format, Uint8 src_channels, int src_rate, + Uint16 dst_format, Uint8 dst_channels, int dst_rate) +{ +/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n", + src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/ + /* Start off with no conversion necessary */ + cvt->needed = 0; + cvt->filter_index = 0; + cvt->filters[0] = NULL; + cvt->len_mult = 1; + cvt->len_ratio = 1.0; + + /* First filter: Endian conversion from src to dst */ + if ( (src_format & 0x1000) != (dst_format & 0x1000) + && ((src_format & 0xff) == 16) && ((dst_format & 0xff) == 16)) { + cvt->filters[cvt->filter_index++] = SDL_ConvertEndian; + } + + /* Second filter: Sign conversion -- signed/unsigned */ + if ( (src_format & 0x8000) != (dst_format & 0x8000) ) { + cvt->filters[cvt->filter_index++] = SDL_ConvertSign; + } + + /* Next filter: Convert 16 bit <--> 8 bit PCM */ + if ( (src_format & 0xFF) != (dst_format & 0xFF) ) { + switch (dst_format&0x10FF) { + case AUDIO_U8: + cvt->filters[cvt->filter_index++] = + SDL_Convert8; + cvt->len_ratio /= 2; + break; + case AUDIO_U16LSB: + cvt->filters[cvt->filter_index++] = + SDL_Convert16LSB; + cvt->len_mult *= 2; + cvt->len_ratio *= 2; + break; + case AUDIO_U16MSB: + cvt->filters[cvt->filter_index++] = + SDL_Convert16MSB; + cvt->len_mult *= 2; + cvt->len_ratio *= 2; + break; + } + } + + /* Last filter: Mono/Stereo conversion */ + if ( src_channels != dst_channels ) { + if ( (src_channels == 1) && (dst_channels > 1) ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertStereo; + cvt->len_mult *= 2; + src_channels = 2; + cvt->len_ratio *= 2; + } + if ( (src_channels == 2) && + (dst_channels == 6) ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertSurround; + src_channels = 6; + cvt->len_mult *= 3; + cvt->len_ratio *= 3; + } + if ( (src_channels == 2) && + (dst_channels == 4) ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertSurround_4; + src_channels = 4; + cvt->len_mult *= 2; + cvt->len_ratio *= 2; + } + while ( (src_channels*2) <= dst_channels ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertStereo; + cvt->len_mult *= 2; + src_channels *= 2; + cvt->len_ratio *= 2; + } + if ( (src_channels == 6) && + (dst_channels <= 2) ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertStrip; + src_channels = 2; + cvt->len_ratio /= 3; + } + if ( (src_channels == 6) && + (dst_channels == 4) ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertStrip_2; + src_channels = 4; + cvt->len_ratio /= 2; + } + /* This assumes that 4 channel audio is in the format: + Left {front/back} + Right {front/back} + so converting to L/R stereo works properly. + */ + while ( ((src_channels%2) == 0) && + ((src_channels/2) >= dst_channels) ) { + cvt->filters[cvt->filter_index++] = + SDL_ConvertMono; + src_channels /= 2; + cvt->len_ratio /= 2; + } + if ( src_channels != dst_channels ) { + /* Uh oh.. */; + } + } + + /* Do rate conversion */ + cvt->rate_incr = 0.0; + if ( (src_rate/100) != (dst_rate/100) ) { + Uint32 hi_rate, lo_rate; + int len_mult; + double len_ratio; + void (SDLCALL *rate_cvt)(SDL_AudioCVT *cvt, Uint16 format); + + if ( src_rate > dst_rate ) { + hi_rate = src_rate; + lo_rate = dst_rate; + switch (src_channels) { + case 1: rate_cvt = SDL_RateDIV2; break; + case 2: rate_cvt = SDL_RateDIV2_c2; break; + case 4: rate_cvt = SDL_RateDIV2_c4; break; + case 6: rate_cvt = SDL_RateDIV2_c6; break; + default: return -1; + } + len_mult = 1; + len_ratio = 0.5; + } else { + hi_rate = dst_rate; + lo_rate = src_rate; + switch (src_channels) { + case 1: rate_cvt = SDL_RateMUL2; break; + case 2: rate_cvt = SDL_RateMUL2_c2; break; + case 4: rate_cvt = SDL_RateMUL2_c4; break; + case 6: rate_cvt = SDL_RateMUL2_c6; break; + default: return -1; + } + len_mult = 2; + len_ratio = 2.0; + } + /* If hi_rate = lo_rate*2^x then conversion is easy */ + while ( ((lo_rate*2)/100) <= (hi_rate/100) ) { + cvt->filters[cvt->filter_index++] = rate_cvt; + cvt->len_mult *= len_mult; + lo_rate *= 2; + cvt->len_ratio *= len_ratio; + } + /* We may need a slow conversion here to finish up */ + if ( (lo_rate/100) != (hi_rate/100) ) { +#if 1 + /* The problem with this is that if the input buffer is + say 1K, and the conversion rate is say 1.1, then the + output buffer is 1.1K, which may not be an acceptable + buffer size for the audio driver (not a power of 2) + */ + /* For now, punt and hope the rate distortion isn't great. + */ +#else + if ( src_rate < dst_rate ) { + cvt->rate_incr = (double)lo_rate/hi_rate; + cvt->len_mult *= 2; + cvt->len_ratio /= cvt->rate_incr; + } else { + cvt->rate_incr = (double)hi_rate/lo_rate; + cvt->len_ratio *= cvt->rate_incr; + } + cvt->filters[cvt->filter_index++] = SDL_RateSLOW; +#endif + } + } + + /* Set up the filter information */ + if ( cvt->filter_index != 0 ) { + cvt->needed = 1; + cvt->src_format = src_format; + cvt->dst_format = dst_format; + cvt->len = 0; + cvt->buf = NULL; + cvt->filters[cvt->filter_index] = NULL; + } + return(cvt->needed); +} diff --git a/apps/plugins/sdl/src/audio/SDL_audiodev.c b/apps/plugins/sdl/src/audio/SDL_audiodev.c new file mode 100644 index 0000000000..396156c7db --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_audiodev.c @@ -0,0 +1,179 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Get the name of the audio device we use for output */ + +#if SDL_AUDIO_DRIVER_BSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO + +#include +#include +#include + +#include "SDL_stdinc.h" +#include "SDL_audiodev_c.h" + +#ifndef _PATH_DEV_DSP +#if defined(__NETBSD__) || defined(__OPENBSD__) +#define _PATH_DEV_DSP "/dev/audio" +#else +#define _PATH_DEV_DSP "/dev/dsp" +#endif +#endif +#ifndef _PATH_DEV_DSP24 +#define _PATH_DEV_DSP24 "/dev/sound/dsp" +#endif +#ifndef _PATH_DEV_AUDIO +#define _PATH_DEV_AUDIO "/dev/audio" +#endif + + +int SDL_OpenAudioPath(char *path, int maxlen, int flags, int classic) +{ + const char *audiodev; + int audio_fd; + char audiopath[1024]; + + /* Figure out what our audio device is */ + if ( ((audiodev=SDL_getenv("SDL_PATH_DSP")) == NULL) && + ((audiodev=SDL_getenv("AUDIODEV")) == NULL) ) { + if ( classic ) { + audiodev = _PATH_DEV_AUDIO; + } else { + struct stat sb; + + /* Added support for /dev/sound/\* in Linux 2.4 */ + if ( ((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode)) && + ((stat(_PATH_DEV_DSP24, &sb) == 0) && S_ISCHR(sb.st_mode)) ) { + audiodev = _PATH_DEV_DSP24; + } else { + audiodev = _PATH_DEV_DSP; + } + } + } + audio_fd = open(audiodev, flags, 0); + + /* If the first open fails, look for other devices */ + if ( (audio_fd < 0) && (SDL_strlen(audiodev) < (sizeof(audiopath)-3)) ) { + int exists, instance; + struct stat sb; + + instance = 1; + do { /* Don't use errno ENOENT - it may not be thread-safe */ + SDL_snprintf(audiopath, SDL_arraysize(audiopath), + "%s%d", audiodev, instance++); + exists = 0; + if ( stat(audiopath, &sb) == 0 ) { + exists = 1; + audio_fd = open(audiopath, flags, 0); + } + } while ( exists && (audio_fd < 0) ); + audiodev = audiopath; + } + if ( path != NULL ) { + SDL_strlcpy(path, audiodev, maxlen); + path[maxlen-1] = '\0'; + } + return(audio_fd); +} + +#elif SDL_AUDIO_DRIVER_PAUD + +/* Get the name of the audio device we use for output */ + +#include +#include + +#include "SDL_stdinc.h" +#include "SDL_audiodev_c.h" + +#ifndef _PATH_DEV_DSP +#define _PATH_DEV_DSP "/dev/%caud%c/%c" +#endif + +char devsettings[][3] = +{ + { 'p', '0', '1' }, { 'p', '0', '2' }, { 'p', '0', '3' }, { 'p', '0', '4' }, + { 'p', '1', '1' }, { 'p', '1', '2' }, { 'p', '1', '3' }, { 'p', '1', '4' }, + { 'p', '2', '1' }, { 'p', '2', '2' }, { 'p', '2', '3' }, { 'p', '2', '4' }, + { 'p', '3', '1' }, { 'p', '3', '2' }, { 'p', '3', '3' }, { 'p', '3', '4' }, + { 'b', '0', '1' }, { 'b', '0', '2' }, { 'b', '0', '3' }, { 'b', '0', '4' }, + { 'b', '1', '1' }, { 'b', '1', '2' }, { 'b', '1', '3' }, { 'b', '1', '4' }, + { 'b', '2', '1' }, { 'b', '2', '2' }, { 'b', '2', '3' }, { 'b', '2', '4' }, + { 'b', '3', '1' }, { 'b', '3', '2' }, { 'b', '3', '3' }, { 'b', '3', '4' }, + { '\0', '\0', '\0' } +}; + +static int OpenUserDefinedDevice(char *path, int maxlen, int flags) +{ + const char *audiodev; + int audio_fd; + + /* Figure out what our audio device is */ + if ((audiodev=SDL_getenv("SDL_PATH_DSP")) == NULL) { + audiodev=SDL_getenv("AUDIODEV"); + } + if ( audiodev == NULL ) { + return -1; + } + audio_fd = open(audiodev, flags, 0); + if ( path != NULL ) { + SDL_strlcpy(path, audiodev, maxlen); + path[maxlen-1] = '\0'; + } + return audio_fd; +} + +int SDL_OpenAudioPath(char *path, int maxlen, int flags, int classic) +{ + struct stat sb; + int audio_fd; + char audiopath[1024]; + int cycle; + + audio_fd = OpenUserDefinedDevice(path,maxlen,flags); + if ( audio_fd != -1 ) { + return audio_fd; + } + + cycle = 0; + while( devsettings[cycle][0] != '\0' ) { + SDL_snprintf( audiopath, SDL_arraysize(audiopath), + _PATH_DEV_DSP, + devsettings[cycle][0], + devsettings[cycle][1], + devsettings[cycle][2]); + + if ( stat(audiopath, &sb) == 0 ) { + audio_fd = open(audiopath, flags, 0); + if ( audio_fd > 0 ) { + if ( path != NULL ) { + SDL_strlcpy( path, audiopath, maxlen ); + } + return audio_fd; + } + } + } + return -1; +} + +#endif /* Audio driver selection */ diff --git a/apps/plugins/sdl/src/audio/SDL_audiodev_c.h b/apps/plugins/sdl/src/audio/SDL_audiodev_c.h new file mode 100644 index 0000000000..179d0d019f --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_audiodev_c.h @@ -0,0 +1,26 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Open the audio device, storing the pathname in 'path' */ +extern int SDL_OpenAudioPath(char *path, int maxlen, int flags, int classic); + diff --git a/apps/plugins/sdl/src/audio/SDL_audiomem.h b/apps/plugins/sdl/src/audio/SDL_audiomem.h new file mode 100644 index 0000000000..a473a5e01d --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_audiomem.h @@ -0,0 +1,25 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#define SDL_AllocAudioMem SDL_malloc +#define SDL_FreeAudioMem SDL_free diff --git a/apps/plugins/sdl/src/audio/SDL_mixer.c b/apps/plugins/sdl/src/audio/SDL_mixer.c new file mode 100644 index 0000000000..b5d4d8bf80 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer.c @@ -0,0 +1,264 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* This provides the default mixing callback for the SDL audio routines */ + +#include "SDL_cpuinfo.h" +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "SDL_sysaudio.h" +#include "SDL_mixer_MMX.h" +#include "SDL_mixer_MMX_VC.h" +#include "SDL_mixer_m68k.h" + +/* This table is used to add two sound values together and pin + * the value to avoid overflow. (used with permission from ARDI) + * Changed to use 0xFE instead of 0xFF for better sound quality. + */ +static const Uint8 mix8[] = +{ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03, + 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, + 0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19, + 0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24, + 0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F, + 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A, + 0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45, + 0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50, + 0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B, + 0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66, + 0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71, + 0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C, + 0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87, + 0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92, + 0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D, + 0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8, + 0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3, + 0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE, + 0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9, + 0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4, + 0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF, + 0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA, + 0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5, + 0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, + 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE +}; + +/* The volume ranges from 0 - 128 */ +#define ADJUST_VOLUME(s, v) (s = (s*v)/SDL_MIX_MAXVOLUME) +#define ADJUST_VOLUME_U8(s, v) (s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128) + +void SDL_MixAudio (Uint8 *dst, const Uint8 *src, Uint32 len, int volume) +{ + Uint16 format; + + if ( volume == 0 ) { + return; + } + /* Mix the user-level audio format */ + if ( current_audio ) { + if ( current_audio->convert.needed ) { + format = current_audio->convert.src_format; + } else { + format = current_audio->spec.format; + } + } else { + /* HACK HACK HACK */ + format = AUDIO_S16; + } + switch (format) { + + case AUDIO_U8: { +#if defined(__GNUC__) && (defined(__m68k__) && !defined(__mcoldfire__)) && defined(SDL_ASSEMBLY_ROUTINES) + SDL_MixAudio_m68k_U8((char*)dst,(char*)src,(unsigned long)len,(long)volume,(char *)mix8); +#else + Uint8 src_sample; + + while ( len-- ) { + src_sample = *src; + ADJUST_VOLUME_U8(src_sample, volume); + *dst = mix8[*dst+src_sample]; + ++dst; + ++src; + } +#endif + } + break; + + case AUDIO_S8: { +#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */ +#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES) + if (SDL_HasMMX()) + { + SDL_MixAudio_MMX_S8((char*)dst,(char*)src,(unsigned int)len,(int)volume); + } + else +#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES) + if (SDL_HasMMX()) + { + SDL_MixAudio_MMX_S8_VC((char*)dst,(char*)src,(unsigned int)len,(int)volume); + } + else +#endif +#endif + +#if defined(__GNUC__) && (defined(__m68k__) && !defined(__mcoldfire__)) && defined(SDL_ASSEMBLY_ROUTINES) + SDL_MixAudio_m68k_S8((char*)dst,(char*)src,(unsigned long)len,(long)volume); +#else + { + Sint8 *dst8, *src8; + Sint8 src_sample; + int dst_sample; + const int max_audioval = ((1<<(8-1))-1); + const int min_audioval = -(1<<(8-1)); + + src8 = (Sint8 *)src; + dst8 = (Sint8 *)dst; + while ( len-- ) { + src_sample = *src8; + ADJUST_VOLUME(src_sample, volume); + dst_sample = *dst8 + src_sample; + if ( dst_sample > max_audioval ) { + *dst8 = max_audioval; + } else + if ( dst_sample < min_audioval ) { + *dst8 = min_audioval; + } else { + *dst8 = dst_sample; + } + ++dst8; + ++src8; + } + } +#endif + } + break; + + case AUDIO_S16LSB: { +#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */ +#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES) + if (SDL_HasMMX()) + { + SDL_MixAudio_MMX_S16((char*)dst,(char*)src,(unsigned int)len,(int)volume); + } + else +#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES) + if (SDL_HasMMX()) + { + SDL_MixAudio_MMX_S16_VC((char*)dst,(char*)src,(unsigned int)len,(int)volume); + } + else +#endif +#endif + +#if defined(__GNUC__) && (defined(__m68k__) && !defined(__mcoldfire__)) && defined(SDL_ASSEMBLY_ROUTINES) + SDL_MixAudio_m68k_S16LSB((short*)dst,(short*)src,(unsigned long)len,(long)volume); +#else + { + Sint16 src1, src2; + int dst_sample; + const int max_audioval = ((1<<(16-1))-1); + const int min_audioval = -(1<<(16-1)); + + len /= 2; + while ( len-- ) { + src1 = ((src[1])<<8|src[0]); + ADJUST_VOLUME(src1, volume); + src2 = ((dst[1])<<8|dst[0]); + src += 2; + dst_sample = src1+src2; + if ( dst_sample > max_audioval ) { + dst_sample = max_audioval; + } else + if ( dst_sample < min_audioval ) { + dst_sample = min_audioval; + } + dst[0] = dst_sample&0xFF; + dst_sample >>= 8; + dst[1] = dst_sample&0xFF; + dst += 2; + } + } +#endif + } + break; + + case AUDIO_S16MSB: { +#if defined(__GNUC__) && (defined(__m68k__) && !defined(__mcoldfire__)) && defined(SDL_ASSEMBLY_ROUTINES) + SDL_MixAudio_m68k_S16MSB((short*)dst,(short*)src,(unsigned long)len,(long)volume); +#else + Sint16 src1, src2; + int dst_sample; + const int max_audioval = ((1<<(16-1))-1); + const int min_audioval = -(1<<(16-1)); + + len /= 2; + while ( len-- ) { + src1 = ((src[0])<<8|src[1]); + ADJUST_VOLUME(src1, volume); + src2 = ((dst[0])<<8|dst[1]); + src += 2; + dst_sample = src1+src2; + if ( dst_sample > max_audioval ) { + dst_sample = max_audioval; + } else + if ( dst_sample < min_audioval ) { + dst_sample = min_audioval; + } + dst[1] = dst_sample&0xFF; + dst_sample >>= 8; + dst[0] = dst_sample&0xFF; + dst += 2; + } +#endif + } + break; + + default: /* If this happens... FIXME! */ + SDL_SetError("SDL_MixAudio(): unknown audio format"); + return; + } +} + diff --git a/apps/plugins/sdl/src/audio/SDL_mixer_MMX.c b/apps/plugins/sdl/src/audio/SDL_mixer_MMX.c new file mode 100644 index 0000000000..a2f1d8dfce --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer_MMX.c @@ -0,0 +1,207 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MMX assembler version of SDL_MixAudio for signed little endian 16 bit samples and signed 8 bit samples + Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr) + This code is licensed under the LGPL (see COPYING for details) + + Assumes buffer size in bytes is a multiple of 16 + Assumes SDL_MIX_MAXVOLUME = 128 +*/ + + +/*********************************************** +* Mixing for 16 bit signed buffers +***********************************************/ + +#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */ +#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES) +void SDL_MixAudio_MMX_S16(char* dst,char* src,unsigned int size,int volume) +{ + __asm__ __volatile__ ( + +" movl %3,%%eax\n" /* eax = volume */ + +" movl %2,%%edx\n" /* edx = size */ + +" shrl $4,%%edx\n" /* process 16 bytes per iteration = 8 samples */ + +" jz .endS16\n" + +" pxor %%mm0,%%mm0\n" + +" movd %%eax,%%mm0\n" +" movq %%mm0,%%mm1\n" +" psllq $16,%%mm0\n" +" por %%mm1,%%mm0\n" +" psllq $16,%%mm0\n" +" por %%mm1,%%mm0\n" +" psllq $16,%%mm0\n" +" por %%mm1,%%mm0\n" /* mm0 = vol|vol|vol|vol */ + +".align 8\n" +" .mixloopS16:\n" + +" movq (%1),%%mm1\n" /* mm1 = a|b|c|d */ + +" movq %%mm1,%%mm2\n" /* mm2 = a|b|c|d */ + +" movq 8(%1),%%mm4\n" /* mm4 = e|f|g|h */ + + /* pré charger le buffer dst dans mm7 */ +" movq (%0),%%mm7\n" /* mm7 = dst[0] */ + + /* multiplier par le volume */ +" pmullw %%mm0,%%mm1\n" /* mm1 = l(a*v)|l(b*v)|l(c*v)|l(d*v) */ + +" pmulhw %%mm0,%%mm2\n" /* mm2 = h(a*v)|h(b*v)|h(c*v)|h(d*v) */ +" movq %%mm4,%%mm5\n" /* mm5 = e|f|g|h */ + +" pmullw %%mm0,%%mm4\n" /* mm4 = l(e*v)|l(f*v)|l(g*v)|l(h*v) */ + +" pmulhw %%mm0,%%mm5\n" /* mm5 = h(e*v)|h(f*v)|h(g*v)|h(h*v) */ +" movq %%mm1,%%mm3\n" /* mm3 = l(a*v)|l(b*v)|l(c*v)|l(d*v) */ + +" punpckhwd %%mm2,%%mm1\n" /* mm1 = a*v|b*v */ + +" movq %%mm4,%%mm6\n" /* mm6 = l(e*v)|l(f*v)|l(g*v)|l(h*v) */ +" punpcklwd %%mm2,%%mm3\n" /* mm3 = c*v|d*v */ + +" punpckhwd %%mm5,%%mm4\n" /* mm4 = e*f|f*v */ + +" punpcklwd %%mm5,%%mm6\n" /* mm6 = g*v|h*v */ + + /* pré charger le buffer dst dans mm5 */ +" movq 8(%0),%%mm5\n" /* mm5 = dst[1] */ + + /* diviser par 128 */ +" psrad $7,%%mm1\n" /* mm1 = a*v/128|b*v/128 , 128 = SDL_MIX_MAXVOLUME */ +" add $16,%1\n" + +" psrad $7,%%mm3\n" /* mm3 = c*v/128|d*v/128 */ + +" psrad $7,%%mm4\n" /* mm4 = e*v/128|f*v/128 */ + + /* mm1 = le sample avec le volume modifié */ +" packssdw %%mm1,%%mm3\n" /* mm3 = s(a*v|b*v|c*v|d*v) */ + +" psrad $7,%%mm6\n" /* mm6= g*v/128|h*v/128 */ +" paddsw %%mm7,%%mm3\n" /* mm3 = adjust_volume(src)+dst */ + + /* mm4 = le sample avec le volume modifié */ +" packssdw %%mm4,%%mm6\n" /* mm6 = s(e*v|f*v|g*v|h*v) */ +" movq %%mm3,(%0)\n" + +" paddsw %%mm5,%%mm6\n" /* mm6 = adjust_volume(src)+dst */ + +" movq %%mm6,8(%0)\n" + +" add $16,%0\n" + +" dec %%edx\n" + +" jnz .mixloopS16\n" + +" emms\n" + +".endS16:\n" + : + : "r" (dst), "r"(src),"m"(size), + "m"(volume) + : "eax","edx","memory" + ); +} + + + +/*////////////////////////////////////////////// */ +/* Mixing for 8 bit signed buffers */ +/*////////////////////////////////////////////// */ + +void SDL_MixAudio_MMX_S8(char* dst,char* src,unsigned int size,int volume) +{ + __asm__ __volatile__ ( + +" movl %3,%%eax\n" /* eax = volume */ + +" movd %%eax,%%mm0\n" +" movq %%mm0,%%mm1\n" +" psllq $16,%%mm0\n" +" por %%mm1,%%mm0\n" +" psllq $16,%%mm0\n" +" por %%mm1,%%mm0\n" +" psllq $16,%%mm0\n" +" por %%mm1,%%mm0\n" + +" movl %2,%%edx\n" /* edx = size */ +" shr $3,%%edx\n" /* process 8 bytes per iteration = 8 samples */ + +" cmp $0,%%edx\n" +" je .endS8\n" + +".align 8\n" +" .mixloopS8:\n" + +" pxor %%mm2,%%mm2\n" /* mm2 = 0 */ +" movq (%1),%%mm1\n" /* mm1 = a|b|c|d|e|f|g|h */ + +" movq %%mm1,%%mm3\n" /* mm3 = a|b|c|d|e|f|g|h */ + + /* on va faire le "sign extension" en faisant un cmp avec 0 qui retourne 1 si <0, 0 si >0 */ +" pcmpgtb %%mm1,%%mm2\n" /* mm2 = 11111111|00000000|00000000.... */ + +" punpckhbw %%mm2,%%mm1\n" /* mm1 = 0|a|0|b|0|c|0|d */ + +" punpcklbw %%mm2,%%mm3\n" /* mm3 = 0|e|0|f|0|g|0|h */ +" movq (%0),%%mm2\n" /* mm2 = destination */ + +" pmullw %%mm0,%%mm1\n" /* mm1 = v*a|v*b|v*c|v*d */ +" add $8,%1\n" + +" pmullw %%mm0,%%mm3\n" /* mm3 = v*e|v*f|v*g|v*h */ +" psraw $7,%%mm1\n" /* mm1 = v*a/128|v*b/128|v*c/128|v*d/128 */ + +" psraw $7,%%mm3\n" /* mm3 = v*e/128|v*f/128|v*g/128|v*h/128 */ + +" packsswb %%mm1,%%mm3\n" /* mm1 = v*a/128|v*b/128|v*c/128|v*d/128|v*e/128|v*f/128|v*g/128|v*h/128 */ + +" paddsb %%mm2,%%mm3\n" /* add to destination buffer */ + +" movq %%mm3,(%0)\n" /* store back to ram */ +" add $8,%0\n" + +" dec %%edx\n" + +" jnz .mixloopS8\n" + +".endS8:\n" +" emms\n" + : + : "r" (dst), "r"(src),"m"(size), + "m"(volume) + : "eax","edx","memory" + ); +} +#endif +#endif diff --git a/apps/plugins/sdl/src/audio/SDL_mixer_MMX.h b/apps/plugins/sdl/src/audio/SDL_mixer_MMX.h new file mode 100644 index 0000000000..836b259592 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer_MMX.h @@ -0,0 +1,17 @@ +/* + headers for MMX assembler version of SDL_MixAudio + Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr) + This code is licensed under the LGPL (see COPYING for details) + + Assumes buffer size in bytes is a multiple of 16 + Assumes SDL_MIX_MAXVOLUME = 128 +*/ +#include "SDL_config.h" + +#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */ +#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES) +void SDL_MixAudio_MMX_S16(char* ,char* ,unsigned int ,int ); +void SDL_MixAudio_MMX_S8(char* ,char* ,unsigned int ,int ); +#endif +#endif + diff --git a/apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.c b/apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.c new file mode 100644 index 0000000000..cd0aa4ccd4 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.c @@ -0,0 +1,183 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#include "SDL_mixer_MMX_VC.h" + +#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */ +#if ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES) +// MMX assembler version of SDL_MixAudio for signed little endian 16 bit samples and signed 8 bit samples +// Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr) +// Converted to Intel ASM notation by Cth +// This code is licensed under the LGPL (see COPYING for details) +// +// Assumes buffer size in bytes is a multiple of 16 +// Assumes SDL_MIX_MAXVOLUME = 128 + + +//////////////////////////////////////////////// +// Mixing for 16 bit signed buffers +//////////////////////////////////////////////// + +void SDL_MixAudio_MMX_S16_VC(char* dst,char* src,unsigned int nSize,int volume) +{ + __asm + { + + push edi + push esi + push ebx + + mov edi, dst // edi = dst + mov esi, src // esi = src + mov eax, volume // eax = volume + mov ebx, nSize // ebx = size + shr ebx, 4 // process 16 bytes per iteration = 8 samples + jz endS16 + + pxor mm0, mm0 + movd mm0, eax //%%eax,%%mm0 + movq mm1, mm0 //%%mm0,%%mm1 + psllq mm0, 16 //$16,%%mm0 + por mm0, mm1 //%%mm1,%%mm0 + psllq mm0, 16 //$16,%%mm0 + por mm0, mm1 //%%mm1,%%mm0 + psllq mm0, 16 //$16,%%mm0 + por mm0, mm1 //%%mm1,%%mm0 // mm0 = vol|vol|vol|vol + + #ifndef __WATCOMC__ + align 16 + #endif +mixloopS16: + movq mm1, [esi] //(%%esi),%%mm1\n" // mm1 = a|b|c|d + movq mm2, mm1 //%%mm1,%%mm2\n" // mm2 = a|b|c|d + movq mm4, [esi + 8] //8(%%esi),%%mm4\n" // mm4 = e|f|g|h + // pre charger le buffer dst dans mm7 + movq mm7, [edi] //(%%edi),%%mm7\n" // mm7 = dst[0]" + // multiplier par le volume + pmullw mm1, mm0 //%%mm0,%%mm1\n" // mm1 = l(a*v)|l(b*v)|l(c*v)|l(d*v) + pmulhw mm2, mm0 //%%mm0,%%mm2\n" // mm2 = h(a*v)|h(b*v)|h(c*v)|h(d*v) + movq mm5, mm4 //%%mm4,%%mm5\n" // mm5 = e|f|g|h + pmullw mm4, mm0 //%%mm0,%%mm4\n" // mm4 = l(e*v)|l(f*v)|l(g*v)|l(h*v) + pmulhw mm5, mm0 //%%mm0,%%mm5\n" // mm5 = h(e*v)|h(f*v)|h(g*v)|h(h*v) + movq mm3, mm1 //%%mm1,%%mm3\n" // mm3 = l(a*v)|l(b*v)|l(c*v)|l(d*v) + punpckhwd mm1, mm2 //%%mm2,%%mm1\n" // mm1 = a*v|b*v + movq mm6, mm4 //%%mm4,%%mm6\n" // mm6 = l(e*v)|l(f*v)|l(g*v)|l(h*v) + punpcklwd mm3, mm2 //%%mm2,%%mm3\n" // mm3 = c*v|d*v + punpckhwd mm4, mm5 //%%mm5,%%mm4\n" // mm4 = e*f|f*v + punpcklwd mm6, mm5 //%%mm5,%%mm6\n" // mm6 = g*v|h*v + // pre charger le buffer dst dans mm5 + movq mm5, [edi + 8] //8(%%edi),%%mm5\n" // mm5 = dst[1] + // diviser par 128 + psrad mm1, 7 //$7,%%mm1\n" // mm1 = a*v/128|b*v/128 , 128 = SDL_MIX_MAXVOLUME + add esi, 16 //$16,%%esi\n" + psrad mm3, 7 //$7,%%mm3\n" // mm3 = c*v/128|d*v/128 + psrad mm4, 7 //$7,%%mm4\n" // mm4 = e*v/128|f*v/128 + // mm1 = le sample avec le volume modifie + packssdw mm3, mm1 //%%mm1,%%mm3\n" // mm3 = s(a*v|b*v|c*v|d*v) + psrad mm6, 7 //$7,%%mm6\n" // mm6= g*v/128|h*v/128 + paddsw mm3, mm7 //%%mm7,%%mm3\n" // mm3 = adjust_volume(src)+dst + // mm4 = le sample avec le volume modifie + packssdw mm6, mm4 //%%mm4,%%mm6\n" // mm6 = s(e*v|f*v|g*v|h*v) + movq [edi], mm3 //%%mm3,(%%edi)\n" + paddsw mm6, mm5 //%%mm5,%%mm6\n" // mm6 = adjust_volume(src)+dst + movq [edi + 8], mm6 //%%mm6,8(%%edi)\n" + add edi, 16 //$16,%%edi\n" + dec ebx //%%ebx\n" + jnz mixloopS16 + +endS16: + emms + + pop ebx + pop esi + pop edi + } + +} + +//////////////////////////////////////////////// +// Mixing for 8 bit signed buffers +//////////////////////////////////////////////// + +void SDL_MixAudio_MMX_S8_VC(char* dst,char* src,unsigned int nSize,int volume) +{ + _asm + { + + push edi + push esi + push ebx + + mov edi, dst //movl %0,%%edi // edi = dst + mov esi, src //%1,%%esi // esi = src + mov eax, volume //%3,%%eax // eax = volume + + movd mm0, eax //%%eax,%%mm0 + movq mm1, mm0 //%%mm0,%%mm1 + psllq mm0, 16 //$16,%%mm0 + por mm0, mm1 //%%mm1,%%mm0 + psllq mm0, 16 //$16,%%mm0 + por mm0, mm1 //%%mm1,%%mm0 + psllq mm0, 16 //$16,%%mm0 + por mm0, mm1 //%%mm1,%%mm0 + + mov ebx, nSize //%2,%%ebx // ebx = size + shr ebx, 3 //$3,%%ebx // process 8 bytes per iteration = 8 samples + cmp ebx, 0 //$0,%%ebx + je endS8 + + #ifndef __WATCOMC__ + align 16 + #endif +mixloopS8: + pxor mm2, mm2 //%%mm2,%%mm2 // mm2 = 0 + movq mm1, [esi] //(%%esi),%%mm1 // mm1 = a|b|c|d|e|f|g|h + movq mm3, mm1 //%%mm1,%%mm3 // mm3 = a|b|c|d|e|f|g|h + // on va faire le "sign extension" en faisant un cmp avec 0 qui retourne 1 si <0, 0 si >0 + pcmpgtb mm2, mm1 //%%mm1,%%mm2 // mm2 = 11111111|00000000|00000000.... + punpckhbw mm1, mm2 //%%mm2,%%mm1 // mm1 = 0|a|0|b|0|c|0|d + punpcklbw mm3, mm2 //%%mm2,%%mm3 // mm3 = 0|e|0|f|0|g|0|h + movq mm2, [edi] //(%%edi),%%mm2 // mm2 = destination + pmullw mm1, mm0 //%%mm0,%%mm1 // mm1 = v*a|v*b|v*c|v*d + add esi, 8 //$8,%%esi + pmullw mm3, mm0 //%%mm0,%%mm3 // mm3 = v*e|v*f|v*g|v*h + psraw mm1, 7 //$7,%%mm1 // mm1 = v*a/128|v*b/128|v*c/128|v*d/128 + psraw mm3, 7 //$7,%%mm3 // mm3 = v*e/128|v*f/128|v*g/128|v*h/128 + packsswb mm3, mm1 //%%mm1,%%mm3 // mm1 = v*a/128|v*b/128|v*c/128|v*d/128|v*e/128|v*f/128|v*g/128|v*h/128 + paddsb mm3, mm2 //%%mm2,%%mm3 // add to destination buffer + movq [edi], mm3 //%%mm3,(%%edi) // store back to ram + add edi, 8 //$8,%%edi + dec ebx //%%ebx + jnz mixloopS8 + +endS8: + emms + + pop ebx + pop esi + pop edi + } +} + +#endif /* SDL_ASSEMBLY_ROUTINES */ +#endif /* SDL_BUGGY_MMX_MIXERS */ diff --git a/apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.h b/apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.h new file mode 100644 index 0000000000..fbd0eb8183 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer_MMX_VC.h @@ -0,0 +1,38 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + + +#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */ +#if ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES) +/* headers for MMX assembler version of SDL_MixAudio + Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr) + Converted to Intel ASM notation by Cth + This code is licensed under the LGPL (see COPYING for details) + + Assumes buffer size in bytes is a multiple of 16 + Assumes SDL_MIX_MAXVOLUME = 128 +*/ +void SDL_MixAudio_MMX_S16_VC(char* ,char* ,unsigned int ,int ); +void SDL_MixAudio_MMX_S8_VC(char* ,char* ,unsigned int ,int ); +#endif +#endif diff --git a/apps/plugins/sdl/src/audio/SDL_mixer_m68k.c b/apps/plugins/sdl/src/audio/SDL_mixer_m68k.c new file mode 100644 index 0000000000..22bb1cb81d --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer_m68k.c @@ -0,0 +1,210 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + m68k assembly mix routines + + Patrice Mandin +*/ + +#if (defined(__m68k__) && !defined(__mcoldfire__)) && defined(__GNUC__) +void SDL_MixAudio_m68k_U8(char* dst, char* src, long len, long volume, char* mix8) +{ + __asm__ __volatile__ ( + + "tstl %2\n" +" beqs stoploop_u8\n" +"mixloop_u8:\n" + + /* Mix a sample */ + +" moveq #0,%%d0\n" +" moveq #0,%%d1\n" + +" moveb %1@+,%%d0\n" /* d0 = *src++ */ +" sub #128,%%d0\n" /* d0 -= 128 */ +" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */ +" moveb %0@,%%d1\n" /* d1 = *dst */ +" asr #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */ +" add #128,%%d0\n" /* d0 += 128 */ + +" add %%d1,%%d0\n" + +" moveb %4@(%%d0:w),%0@+\n" + + /* Loop till done */ + +" subql #1,%2\n" +" bhis mixloop_u8\n" +"stoploop_u8:\n" + + : /* no return value */ + : /* input */ + "a"(dst), "a"(src), "d"(len), "d"(volume), "a"(mix8) + : /* clobbered registers */ + "d0", "d1", "cc", "memory" + ); +} + +void SDL_MixAudio_m68k_S8(char* dst, char* src, long len, long volume) +{ + __asm__ __volatile__ ( + + "tstl %2\n" +" beqs stoploop_s8\n" +" moveq #-128,%%d2\n" +" moveq #127,%%d3\n" +"mixloop_s8:\n" + + /* Mix a sample */ + +" moveq #0,%%d0\n" +" moveq #0,%%d1\n" + +" moveb %1@+,%%d0\n" /* d0 = *src++ */ +" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */ +" moveb %0@,%%d1\n" /* d1 = *dst */ +" asr #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */ + +" add %%d1,%%d0\n" + +" cmp %%d2,%%d0\n" +" bges lower_limit_s8\n" +" move %%d2,%%d0\n" +"lower_limit_s8:\n" + +" cmp %%d3,%%d0\n" +" bles upper_limit_s8\n" +" move %%d3,%%d0\n" +"upper_limit_s8:\n" +" moveb %%d0,%0@+\n" + + /* Loop till done */ + +" subql #1,%2\n" +" bhis mixloop_s8\n" +"stoploop_s8:\n" + + : /* no return value */ + : /* input */ + "a"(dst), "a"(src), "d"(len), "d"(volume) + : /* clobbered registers */ + "d0", "d1", "d2", "d3", "cc", "memory" + ); +} + +void SDL_MixAudio_m68k_S16MSB(short* dst, short* src, long len, long volume) +{ + __asm__ __volatile__ ( + + "tstl %2\n" +" beqs stoploop_s16msb\n" +" movel #-32768,%%d2\n" +" movel #32767,%%d3\n" +" lsrl #1,%2\n" +"mixloop_s16msb:\n" + + /* Mix a sample */ + +" move %1@+,%%d0\n" /* d0 = *src++ */ +" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */ +" move %0@,%%d1\n" /* d1 = *dst */ +" extl %%d1\n" /* extend d1 to 32 bits */ +" asrl #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */ + +" addl %%d1,%%d0\n" + +" cmpl %%d2,%%d0\n" +" bges lower_limit_s16msb\n" +" move %%d2,%%d0\n" +"lower_limit_s16msb:\n" + +" cmpl %%d3,%%d0\n" +" bles upper_limit_s16msb\n" +" move %%d3,%%d0\n" +"upper_limit_s16msb:\n" +" move %%d0,%0@+\n" + + /* Loop till done */ + +" subql #1,%2\n" +" bhis mixloop_s16msb\n" +"stoploop_s16msb:\n" + + : /* no return value */ + : /* input */ + "a"(dst), "a"(src), "d"(len), "d"(volume) + : /* clobbered registers */ + "d0", "d1", "d2", "d3", "cc", "memory" + ); +} + +void SDL_MixAudio_m68k_S16LSB(short* dst, short* src, long len, long volume) +{ + __asm__ __volatile__ ( + + "tstl %2\n" +" beqs stoploop_s16lsb\n" +" movel #-32768,%%d2\n" +" movel #32767,%%d3\n" +" lsrl #1,%2\n" +"mixloop_s16lsb:\n" + + /* Mix a sample */ + +" move %1@+,%%d0\n" /* d0 = *src++ */ +" rorw #8,%%d0\n" +" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */ +" move %0@,%%d1\n" /* d1 = *dst */ +" rorw #8,%%d1\n" +" extl %%d1\n" /* extend d1 to 32 bits */ +" asrl #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */ + +" addl %%d1,%%d0\n" + +" cmpl %%d2,%%d0\n" +" bges lower_limit_s16lsb\n" +" move %%d2,%%d0\n" +"lower_limit_s16lsb:\n" + +" cmpl %%d3,%%d0\n" +" bles upper_limit_s16lsb\n" +" move %%d3,%%d0\n" +"upper_limit_s16lsb:\n" +" rorw #8,%%d0\n" +" move %%d0,%0@+\n" + + /* Loop till done */ + +" subql #1,%2\n" +" bhis mixloop_s16lsb\n" +"stoploop_s16lsb:\n" + + : /* no return value */ + : /* input */ + "a"(dst), "a"(src), "d"(len), "d"(volume) + : /* clobbered registers */ + "d0", "d1", "d2", "d3", "cc", "memory" + ); +} +#endif diff --git a/apps/plugins/sdl/src/audio/SDL_mixer_m68k.h b/apps/plugins/sdl/src/audio/SDL_mixer_m68k.h new file mode 100644 index 0000000000..673df00656 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_mixer_m68k.h @@ -0,0 +1,36 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + m68k assembly mix routines + + Patrice Mandin +*/ + +#if (defined(__m68k__) && !defined(__mcoldfire__)) && defined(__GNUC__) +void SDL_MixAudio_m68k_U8(char* dst,char* src, long len, long volume, char* mix8); +void SDL_MixAudio_m68k_S8(char* dst,char* src, long len, long volume); + +void SDL_MixAudio_m68k_S16MSB(short* dst,short* src, long len, long volume); +void SDL_MixAudio_m68k_S16LSB(short* dst,short* src, long len, long volume); +#endif diff --git a/apps/plugins/sdl/src/audio/SDL_sysaudio.h b/apps/plugins/sdl/src/audio/SDL_sysaudio.h new file mode 100644 index 0000000000..7060cef4df --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_sysaudio.h @@ -0,0 +1,189 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is SDL_free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_sysaudio_h +#define _SDL_sysaudio_h + +#include "SDL_mutex.h" +#include "SDL_thread.h" + +/* The SDL audio driver */ +typedef struct SDL_AudioDevice SDL_AudioDevice; + +/* Define the SDL audio driver structure */ +#define _THIS SDL_AudioDevice *_this +#ifndef _STATUS +#define _STATUS SDL_status *status +#endif +struct SDL_AudioDevice { + /* * * */ + /* The name of this audio driver */ + const char *name; + + /* * * */ + /* The description of this audio driver */ + const char *desc; + + /* * * */ + /* Public driver functions */ + int (*OpenAudio)(_THIS, SDL_AudioSpec *spec); + void (*ThreadInit)(_THIS); /* Called by audio thread at start */ + void (*WaitAudio)(_THIS); + void (*PlayAudio)(_THIS); + Uint8 *(*GetAudioBuf)(_THIS); + void (*WaitDone)(_THIS); + void (*CloseAudio)(_THIS); + + /* * * */ + /* Lock / Unlock functions added for the Mac port */ + void (*LockAudio)(_THIS); + void (*UnlockAudio)(_THIS); + + void (*SetCaption)(_THIS, const char *caption); + + /* * * */ + /* Data common to all devices */ + + /* The current audio specification (shared with audio thread) */ + SDL_AudioSpec spec; + + /* An audio conversion block for audio format emulation */ + SDL_AudioCVT convert; + + /* Current state flags */ + int enabled; + int paused; + int opened; + + /* Fake audio buffer for when the audio hardware is busy */ + Uint8 *fake_stream; + + /* A semaphore for locking the mixing buffers */ + SDL_mutex *mixer_lock; + + /* A thread to feed the audio device */ + SDL_Thread *thread; + Uint32 threadid; + + /* * * */ + /* Data private to this driver */ + struct SDL_PrivateAudioData *hidden; + + /* * * */ + /* The function used to dispose of this structure */ + void (*free)(_THIS); +}; +#undef _THIS + +typedef struct AudioBootStrap { + const char *name; + const char *desc; + int (*available)(void); + SDL_AudioDevice *(*create)(int devindex); +} AudioBootStrap; + +#if SDL_AUDIO_DRIVER_BSD +extern AudioBootStrap BSD_AUDIO_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_PULSE +extern AudioBootStrap PULSE_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_ALSA +extern AudioBootStrap ALSA_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_OSS +extern AudioBootStrap DSP_bootstrap; +extern AudioBootStrap DMA_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_QNXNTO +extern AudioBootStrap QNXNTOAUDIO_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_SUNAUDIO +extern AudioBootStrap SUNAUDIO_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_DMEDIA +extern AudioBootStrap DMEDIA_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_ARTS +extern AudioBootStrap ARTS_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_ESD +extern AudioBootStrap ESD_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_NAS +extern AudioBootStrap NAS_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_DSOUND +extern AudioBootStrap DSOUND_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_WAVEOUT +extern AudioBootStrap WAVEOUT_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_PAUD +extern AudioBootStrap Paud_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_BAUDIO +extern AudioBootStrap BAUDIO_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_COREAUDIO +extern AudioBootStrap COREAUDIO_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_SNDMGR +extern AudioBootStrap SNDMGR_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_MINT +extern AudioBootStrap MINTAUDIO_GSXB_bootstrap; +extern AudioBootStrap MINTAUDIO_MCSN_bootstrap; +extern AudioBootStrap MINTAUDIO_STFA_bootstrap; +extern AudioBootStrap MINTAUDIO_XBIOS_bootstrap; +extern AudioBootStrap MINTAUDIO_DMA8_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_DISK +extern AudioBootStrap DISKAUD_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_DUMMY +extern AudioBootStrap DUMMYAUD_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_DC +extern AudioBootStrap DCAUD_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_NDS +extern AudioBootStrap NDSAUD_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_MMEAUDIO +extern AudioBootStrap MMEAUDIO_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_DART +extern AudioBootStrap DART_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_EPOCAUDIO +extern AudioBootStrap EPOCAudio_bootstrap; +#endif +#if SDL_AUDIO_DRIVER_ROCKBOX +extern AudioBootStrap ROCKBOXAUD_bootstrap; +#endif + +/* This is the current audio device */ +extern SDL_AudioDevice *current_audio; + +#endif /* _SDL_sysaudio_h */ diff --git a/apps/plugins/sdl/src/audio/SDL_wave.c b/apps/plugins/sdl/src/audio/SDL_wave.c new file mode 100644 index 0000000000..b4ad6c7876 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_wave.c @@ -0,0 +1,596 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Microsoft WAVE file loading routines */ + +#include "SDL_audio.h" +#include "SDL_wave.h" + + +static int ReadChunk(SDL_RWops *src, Chunk *chunk); + +struct MS_ADPCM_decodestate { + Uint8 hPredictor; + Uint16 iDelta; + Sint16 iSamp1; + Sint16 iSamp2; +}; +static struct MS_ADPCM_decoder { + WaveFMT wavefmt; + Uint16 wSamplesPerBlock; + Uint16 wNumCoef; + Sint16 aCoeff[7][2]; + /* * * */ + struct MS_ADPCM_decodestate state[2]; +} MS_ADPCM_state; + +static int InitMS_ADPCM(WaveFMT *format) +{ + Uint8 *rogue_feel; + int i; + + /* Set the rogue pointer to the MS_ADPCM specific data */ + MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); + MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); + MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); + MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); + MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); + MS_ADPCM_state.wavefmt.bitspersample = + SDL_SwapLE16(format->bitspersample); + rogue_feel = (Uint8 *)format+sizeof(*format); + if ( sizeof(*format) == 16 ) { + rogue_feel += sizeof(Uint16); + } + MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); + rogue_feel += sizeof(Uint16); + MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]); + rogue_feel += sizeof(Uint16); + if ( MS_ADPCM_state.wNumCoef != 7 ) { + SDL_SetError("Unknown set of MS_ADPCM coefficients"); + return(-1); + } + for ( i=0; iiSamp1 * coeff[0]) + + (state->iSamp2 * coeff[1]))/256; + if ( nybble & 0x08 ) { + new_sample += state->iDelta * (nybble-0x10); + } else { + new_sample += state->iDelta * nybble; + } + if ( new_sample < min_audioval ) { + new_sample = min_audioval; + } else + if ( new_sample > max_audioval ) { + new_sample = max_audioval; + } + delta = ((Sint32)state->iDelta * adaptive[nybble])/256; + if ( delta < 16 ) { + delta = 16; + } + state->iDelta = (Uint16)delta; + state->iSamp2 = state->iSamp1; + state->iSamp1 = (Sint16)new_sample; + return(new_sample); +} + +static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) +{ + struct MS_ADPCM_decodestate *state[2]; + Uint8 *freeable, *encoded, *decoded; + Sint32 encoded_len, samplesleft; + Sint8 nybble, stereo; + Sint16 *coeff[2]; + Sint32 new_sample; + + /* Allocate the proper sized output buffer */ + encoded_len = *audio_len; + encoded = *audio_buf; + freeable = *audio_buf; + *audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * + MS_ADPCM_state.wSamplesPerBlock* + MS_ADPCM_state.wavefmt.channels*sizeof(Sint16); + *audio_buf = (Uint8 *)SDL_malloc(*audio_len); + if ( *audio_buf == NULL ) { + SDL_Error(SDL_ENOMEM); + return(-1); + } + decoded = *audio_buf; + + /* Get ready... Go! */ + stereo = (MS_ADPCM_state.wavefmt.channels == 2); + state[0] = &MS_ADPCM_state.state[0]; + state[1] = &MS_ADPCM_state.state[stereo]; + while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) { + /* Grab the initial information for this block */ + state[0]->hPredictor = *encoded++; + if ( stereo ) { + state[1]->hPredictor = *encoded++; + } + state[0]->iDelta = ((encoded[1]<<8)|encoded[0]); + encoded += sizeof(Sint16); + if ( stereo ) { + state[1]->iDelta = ((encoded[1]<<8)|encoded[0]); + encoded += sizeof(Sint16); + } + state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]); + encoded += sizeof(Sint16); + if ( stereo ) { + state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]); + encoded += sizeof(Sint16); + } + state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]); + encoded += sizeof(Sint16); + if ( stereo ) { + state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]); + encoded += sizeof(Sint16); + } + coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor]; + coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor]; + + /* Store the two initial samples we start with */ + decoded[0] = state[0]->iSamp2&0xFF; + decoded[1] = state[0]->iSamp2>>8; + decoded += 2; + if ( stereo ) { + decoded[0] = state[1]->iSamp2&0xFF; + decoded[1] = state[1]->iSamp2>>8; + decoded += 2; + } + decoded[0] = state[0]->iSamp1&0xFF; + decoded[1] = state[0]->iSamp1>>8; + decoded += 2; + if ( stereo ) { + decoded[0] = state[1]->iSamp1&0xFF; + decoded[1] = state[1]->iSamp1>>8; + decoded += 2; + } + + /* Decode and store the other samples in this block */ + samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)* + MS_ADPCM_state.wavefmt.channels; + while ( samplesleft > 0 ) { + nybble = (*encoded)>>4; + new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]); + decoded[0] = new_sample&0xFF; + new_sample >>= 8; + decoded[1] = new_sample&0xFF; + decoded += 2; + + nybble = (*encoded)&0x0F; + new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]); + decoded[0] = new_sample&0xFF; + new_sample >>= 8; + decoded[1] = new_sample&0xFF; + decoded += 2; + + ++encoded; + samplesleft -= 2; + } + encoded_len -= MS_ADPCM_state.wavefmt.blockalign; + } + SDL_free(freeable); + return(0); +} + +struct IMA_ADPCM_decodestate { + Sint32 sample; + Sint8 index; +}; +static struct IMA_ADPCM_decoder { + WaveFMT wavefmt; + Uint16 wSamplesPerBlock; + /* * * */ + struct IMA_ADPCM_decodestate state[2]; +} IMA_ADPCM_state; + +static int InitIMA_ADPCM(WaveFMT *format) +{ + Uint8 *rogue_feel; + + /* Set the rogue pointer to the IMA_ADPCM specific data */ + IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding); + IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels); + IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency); + IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate); + IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign); + IMA_ADPCM_state.wavefmt.bitspersample = + SDL_SwapLE16(format->bitspersample); + rogue_feel = (Uint8 *)format+sizeof(*format); + if ( sizeof(*format) == 16 ) { + rogue_feel += sizeof(Uint16); + } + IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]); + return(0); +} + +static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble) +{ + const Sint32 max_audioval = ((1<<(16-1))-1); + const Sint32 min_audioval = -(1<<(16-1)); + const int index_table[16] = { + -1, -1, -1, -1, + 2, 4, 6, 8, + -1, -1, -1, -1, + 2, 4, 6, 8 + }; + const Sint32 step_table[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, + 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, + 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, + 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, + 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, + 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, + 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, + 22385, 24623, 27086, 29794, 32767 + }; + Sint32 delta, step; + + /* Compute difference and new sample value */ + step = step_table[state->index]; + delta = step >> 3; + if ( nybble & 0x04 ) delta += step; + if ( nybble & 0x02 ) delta += (step >> 1); + if ( nybble & 0x01 ) delta += (step >> 2); + if ( nybble & 0x08 ) delta = -delta; + state->sample += delta; + + /* Update index value */ + state->index += index_table[nybble]; + if ( state->index > 88 ) { + state->index = 88; + } else + if ( state->index < 0 ) { + state->index = 0; + } + + /* Clamp output sample */ + if ( state->sample > max_audioval ) { + state->sample = max_audioval; + } else + if ( state->sample < min_audioval ) { + state->sample = min_audioval; + } + return(state->sample); +} + +/* Fill the decode buffer with a channel block of data (8 samples) */ +static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded, + int channel, int numchannels, struct IMA_ADPCM_decodestate *state) +{ + int i; + Sint8 nybble; + Sint32 new_sample; + + decoded += (channel * 2); + for ( i=0; i<4; ++i ) { + nybble = (*encoded)&0x0F; + new_sample = IMA_ADPCM_nibble(state, nybble); + decoded[0] = new_sample&0xFF; + new_sample >>= 8; + decoded[1] = new_sample&0xFF; + decoded += 2 * numchannels; + + nybble = (*encoded)>>4; + new_sample = IMA_ADPCM_nibble(state, nybble); + decoded[0] = new_sample&0xFF; + new_sample >>= 8; + decoded[1] = new_sample&0xFF; + decoded += 2 * numchannels; + + ++encoded; + } +} + +static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len) +{ + struct IMA_ADPCM_decodestate *state; + Uint8 *freeable, *encoded, *decoded; + Sint32 encoded_len, samplesleft; + unsigned int c, channels; + + /* Check to make sure we have enough variables in the state array */ + channels = IMA_ADPCM_state.wavefmt.channels; + if ( channels > SDL_arraysize(IMA_ADPCM_state.state) ) { + SDL_SetError("IMA ADPCM decoder can only handle %d channels", + SDL_arraysize(IMA_ADPCM_state.state)); + return(-1); + } + state = IMA_ADPCM_state.state; + + /* Allocate the proper sized output buffer */ + encoded_len = *audio_len; + encoded = *audio_buf; + freeable = *audio_buf; + *audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * + IMA_ADPCM_state.wSamplesPerBlock* + IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16); + *audio_buf = (Uint8 *)SDL_malloc(*audio_len); + if ( *audio_buf == NULL ) { + SDL_Error(SDL_ENOMEM); + return(-1); + } + decoded = *audio_buf; + + /* Get ready... Go! */ + while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) { + /* Grab the initial information for this block */ + for ( c=0; c>8); + decoded += 2; + } + + /* Decode and store the other samples in this block */ + samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels; + while ( samplesleft > 0 ) { + for ( c=0; cencoding)) { + case PCM_CODE: + /* We can understand this */ + break; + case MS_ADPCM_CODE: + /* Try to understand this */ + if ( InitMS_ADPCM(format) < 0 ) { + was_error = 1; + goto done; + } + MS_ADPCM_encoded = 1; + break; + case IMA_ADPCM_CODE: + /* Try to understand this */ + if ( InitIMA_ADPCM(format) < 0 ) { + was_error = 1; + goto done; + } + IMA_ADPCM_encoded = 1; + break; + case MP3_CODE: + SDL_SetError("MPEG Layer 3 data not supported", + SDL_SwapLE16(format->encoding)); + was_error = 1; + goto done; + default: + SDL_SetError("Unknown WAVE data format: 0x%.4x", + SDL_SwapLE16(format->encoding)); + was_error = 1; + goto done; + } + SDL_memset(spec, 0, (sizeof *spec)); + spec->freq = SDL_SwapLE32(format->frequency); + switch (SDL_SwapLE16(format->bitspersample)) { + case 4: + if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) { + spec->format = AUDIO_S16; + } else { + was_error = 1; + } + break; + case 8: + spec->format = AUDIO_U8; + break; + case 16: + spec->format = AUDIO_S16; + break; + default: + was_error = 1; + break; + } + if ( was_error ) { + SDL_SetError("Unknown %d-bit PCM data format", + SDL_SwapLE16(format->bitspersample)); + goto done; + } + spec->channels = (Uint8)SDL_SwapLE16(format->channels); + spec->samples = 4096; /* Good default buffer size */ + + /* Read the audio data chunk */ + *audio_buf = NULL; + do { + if ( *audio_buf != NULL ) { + SDL_free(*audio_buf); + *audio_buf = NULL; + } + lenread = ReadChunk(src, &chunk); + if ( lenread < 0 ) { + was_error = 1; + goto done; + } + *audio_len = lenread; + *audio_buf = chunk.data; + if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32); + } while ( chunk.magic != DATA ); + headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */ + + if ( MS_ADPCM_encoded ) { + if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) { + was_error = 1; + goto done; + } + } + if ( IMA_ADPCM_encoded ) { + if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) { + was_error = 1; + goto done; + } + } + + /* Don't return a buffer that isn't a multiple of samplesize */ + samplesize = ((spec->format & 0xFF)/8)*spec->channels; + *audio_len &= ~(samplesize-1); + +done: + if ( format != NULL ) { + SDL_free(format); + } + if ( src ) { + if ( freesrc ) { + SDL_RWclose(src); + } else { + /* seek to the end of the file (given by the RIFF chunk) */ + SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR); + } + } + if ( was_error ) { + spec = NULL; + } + return(spec); +} + +/* Since the WAV memory is allocated in the shared library, it must also + be freed here. (Necessary under Win32, VC++) + */ +void SDL_FreeWAV(Uint8 *audio_buf) +{ + if ( audio_buf != NULL ) { + SDL_free(audio_buf); + } +} + +static int ReadChunk(SDL_RWops *src, Chunk *chunk) +{ + chunk->magic = SDL_ReadLE32(src); + chunk->length = SDL_ReadLE32(src); + chunk->data = (Uint8 *)SDL_malloc(chunk->length); + if ( chunk->data == NULL ) { + SDL_Error(SDL_ENOMEM); + return(-1); + } + if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) { + SDL_Error(SDL_EFREAD); + SDL_free(chunk->data); + chunk->data = NULL; + return(-1); + } + return(chunk->length); +} diff --git a/apps/plugins/sdl/src/audio/SDL_wave.h b/apps/plugins/sdl/src/audio/SDL_wave.h new file mode 100644 index 0000000000..53b12e70d2 --- /dev/null +++ b/apps/plugins/sdl/src/audio/SDL_wave.h @@ -0,0 +1,62 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is SDL_free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* WAVE files are little-endian */ + +/*******************************************/ +/* Define values for Microsoft WAVE format */ +/*******************************************/ +#define RIFF 0x46464952 /* "RIFF" */ +#define WAVE 0x45564157 /* "WAVE" */ +#define FACT 0x74636166 /* "fact" */ +#define LIST 0x5453494c /* "LIST" */ +#define FMT 0x20746D66 /* "fmt " */ +#define DATA 0x61746164 /* "data" */ +#define PCM_CODE 0x0001 +#define MS_ADPCM_CODE 0x0002 +#define IMA_ADPCM_CODE 0x0011 +#define MP3_CODE 0x0055 +#define WAVE_MONO 1 +#define WAVE_STEREO 2 + +/* Normally, these three chunks come consecutively in a WAVE file */ +typedef struct WaveFMT { +/* 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; + +/* The general chunk found in the WAVE file */ +typedef struct Chunk { + Uint32 magic; + Uint32 length; + Uint8 *data; +} Chunk; + diff --git a/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c b/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c new file mode 100644 index 0000000000..f10733e432 --- /dev/null +++ b/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c @@ -0,0 +1,619 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include +#include /* For kill() */ + +#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_name.h" +#include "SDL_loadso.h" +#else +#define SDL_NAME(X) X +#endif + + +/* The tag name used by ALSA audio */ +#define DRIVER_NAME "alsa" + +/* Audio driver functions */ +static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void ALSA_WaitAudio(_THIS); +static void ALSA_PlayAudio(_THIS); +static Uint8 *ALSA_GetAudioBuf(_THIS); +static void ALSA_CloseAudio(_THIS); + +#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC + +static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; +static void *alsa_handle = NULL; +static int alsa_loaded = 0; + +static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); +static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); +static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); +static int (*SDL_NAME(snd_pcm_recover))(snd_pcm_t *pcm, int err, int silent); +static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); +static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); +static const char *(*SDL_NAME(snd_strerror))(int errnum); +static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); +static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void); +static void (*SDL_NAME(snd_pcm_hw_params_copy))(snd_pcm_hw_params_t *dst, const snd_pcm_hw_params_t *src); +static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); +static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access); +static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val); +static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); +static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params, unsigned int *val); +static int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +static int (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val, int *dir); +static int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir); +static int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); +static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); +static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); +/* +*/ +static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams, snd_pcm_uframes_t val); +static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams); +static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); +static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params); +static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); +static int (*SDL_NAME(snd_pcm_wait))(snd_pcm_t *pcm, int timeout); +#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) +#define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof) + +/* cast funcs to char* first, to please GCC's strict aliasing rules. */ +static struct { + const char *name; + void **func; +} alsa_functions[] = { + { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, + { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, + { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, + { "snd_pcm_recover", (void**)(char*)&SDL_NAME(snd_pcm_recover) }, + { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, + { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, + { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, + { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, + { "snd_pcm_sw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof) }, + { "snd_pcm_hw_params_copy", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_copy) }, + { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, + { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, + { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, + { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, + { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, + { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, + { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, + { "snd_pcm_hw_params_get_period_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size) }, + { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, + { "snd_pcm_hw_params_get_periods", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods) }, + { "snd_pcm_hw_params_set_buffer_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) }, + { "snd_pcm_hw_params_get_buffer_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) }, + { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, + { "snd_pcm_sw_params_set_avail_min", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_avail_min) }, + { "snd_pcm_sw_params_current", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_current) }, + { "snd_pcm_sw_params_set_start_threshold", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold) }, + { "snd_pcm_sw_params", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) }, + { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, + { "snd_pcm_wait", (void**)(char*)&SDL_NAME(snd_pcm_wait) }, +}; + +static void UnloadALSALibrary(void) { + if (alsa_loaded) { + SDL_UnloadObject(alsa_handle); + alsa_handle = NULL; + alsa_loaded = 0; + } +} + +static int LoadALSALibrary(void) { + int i, retval = -1; + + alsa_handle = SDL_LoadObject(alsa_library); + if (alsa_handle) { + alsa_loaded = 1; + retval = 0; + for (i = 0; i < SDL_arraysize(alsa_functions); i++) { + *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name); + if (!*alsa_functions[i].func) { + retval = -1; + UnloadALSALibrary(); + break; + } + } + } + return retval; +} + +#else + +static void UnloadALSALibrary(void) { + return; +} + +static int LoadALSALibrary(void) { + 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; +} + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + int available; + int status; + snd_pcm_t *handle; + + available = 0; + if (LoadALSALibrary() < 0) { + return available; + } + status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); + if ( status >= 0 ) { + available = 1; + SDL_NAME(snd_pcm_close)(handle); + } + UnloadALSALibrary(); + return(available); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); + UnloadALSALibrary(); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + LoadALSALibrary(); + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = ALSA_OpenAudio; + this->WaitAudio = ALSA_WaitAudio; + this->PlayAudio = ALSA_PlayAudio; + this->GetAudioBuf = ALSA_GetAudioBuf; + this->CloseAudio = ALSA_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap ALSA_bootstrap = { + DRIVER_NAME, "ALSA PCM audio", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void ALSA_WaitAudio(_THIS) +{ + /* We're in blocking mode, so there's nothing to do here */ +} + + +/* + * 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 *) 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 __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); } +static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); } +static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); } +static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); } + +#undef SWIZ6 + + +/* + * Called right before feeding this->mixbuf to the hardware. Swizzle channels + * from Windows/Mac order to the format alsalib will want. + */ +static __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_PlayAudio(_THIS) +{ + int status; + snd_pcm_uframes_t frames_left; + const Uint8 *sample_buf = (const Uint8 *) mixbuf; + const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) * this->spec.channels; + + swizzle_alsa_channels(this); + + frames_left = ((snd_pcm_uframes_t) this->spec.samples); + + while ( frames_left > 0 && this->enabled ) { + /* This works, but needs more testing before going live */ + /*SDL_NAME(snd_pcm_wait)(pcm_handle, -1);*/ + + status = SDL_NAME(snd_pcm_writei)(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 = SDL_NAME(snd_pcm_recover)(pcm_handle, status, 0); + if ( status < 0 ) { + /* Hmm, not much we can do - abort */ + fprintf(stderr, "ALSA write failed (unrecoverable): %s\n", SDL_NAME(snd_strerror)(status)); + this->enabled = 0; + return; + } + continue; + } + sample_buf += status * frame_size; + frames_left -= status; + } +} + +static Uint8 *ALSA_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void ALSA_CloseAudio(_THIS) +{ + if ( mixbuf != NULL ) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if ( pcm_handle ) { + SDL_NAME(snd_pcm_drain)(pcm_handle); + SDL_NAME(snd_pcm_close)(pcm_handle); + pcm_handle = NULL; + } +} + +static int ALSA_finalize_hardware(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *hwparams, int override) +{ + int status; + snd_pcm_uframes_t bufsize; + + /* "set" the hardware with the desired parameters */ + status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams); + if ( status < 0 ) { + return(-1); + } + + /* Get samples for the actual buffer size */ + status = SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize); + if ( status < 0 ) { + return(-1); + } + if ( !override && bufsize != spec->samples * 2 ) { + return(-1); + } + + /* FIXME: Is this safe to do? */ + spec->samples = bufsize / 2; + + /* This is useful for debugging */ + if ( getenv("SDL_AUDIO_ALSA_DEBUG") ) { + snd_pcm_uframes_t persize = 0; + unsigned int periods = 0; + + SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, NULL); + SDL_NAME(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, SDL_AudioSpec *spec, 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); + SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params); + + if ( !override ) { + env = getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE"); + if ( env ) { + override = SDL_atoi(env); + if ( override == 0 ) { + return(-1); + } + } + } + + frames = spec->samples; + status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL); + if ( status < 0 ) { + return(-1); + } + + periods = 2; + status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL); + if ( status < 0 ) { + return(-1); + } + + return ALSA_finalize_hardware(this, spec, hwparams, override); +} + +static int ALSA_set_buffer_size(_THIS, SDL_AudioSpec *spec, 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); + SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params); + + if ( !override ) { + env = getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE"); + if ( env ) { + override = SDL_atoi(env); + if ( override == 0 ) { + return(-1); + } + } + } + + frames = spec->samples * 2; + status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames); + if ( status < 0 ) { + return(-1); + } + + return ALSA_finalize_hardware(this, spec, hwparams, override); +} + +static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + int status; + snd_pcm_hw_params_t *hwparams; + snd_pcm_sw_params_t *swparams; + snd_pcm_format_t format; + unsigned int rate; + unsigned int channels; + Uint16 test_format; + + /* Open the audio device */ + /* Name of device should depend on # channels in spec */ + status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); + + if ( status < 0 ) { + SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); + return(-1); + } + + /* Figure out what the hardware is capable of */ + snd_pcm_hw_params_alloca(&hwparams); + status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams); + if ( status < 0 ) { + SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + + /* SDL only uses interleaved sample output */ + status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); + if ( status < 0 ) { + SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + + /* Try for a closest match on audio format */ + status = -1; + for ( test_format = SDL_FirstAudioFormat(spec->format); + test_format && (status < 0); ) { + 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; + default: + format = 0; + break; + } + if ( format != 0 ) { + status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format); + } + if ( status < 0 ) { + test_format = SDL_NextAudioFormat(); + } + } + if ( status < 0 ) { + SDL_SetError("Couldn't find any hardware audio formats"); + ALSA_CloseAudio(this); + return(-1); + } + spec->format = test_format; + + /* Set the number of channels */ + status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels); + channels = spec->channels; + if ( status < 0 ) { + status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels); + if ( status < 0 ) { + SDL_SetError("Couldn't set audio channels"); + ALSA_CloseAudio(this); + return(-1); + } + spec->channels = channels; + } + + /* Set the audio rate */ + rate = spec->freq; + + status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL); + if ( status < 0 ) { + SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + spec->freq = rate; + + /* Set the buffer size, in samples */ + if ( ALSA_set_period_size(this, spec, hwparams, 0) < 0 && + ALSA_set_buffer_size(this, spec, hwparams, 0) < 0 ) { + /* Failed to set desired buffer size, do the best you can... */ + if ( ALSA_set_period_size(this, spec, hwparams, 1) < 0 ) { + SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + } + + /* Set the software parameters */ + snd_pcm_sw_params_alloca(&swparams); + status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams); + if ( status < 0 ) { + SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, spec->samples); + if ( status < 0 ) { + SDL_SetError("Couldn't set minimum available samples: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 1); + if ( status < 0 ) { + SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams); + if ( status < 0 ) { + SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Allocate mixing buffer */ + mixlen = spec->size; + mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); + if ( mixbuf == NULL ) { + ALSA_CloseAudio(this); + return(-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* Switch to blocking mode for playback */ + SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h b/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h new file mode 100644 index 0000000000..55ae87b8ac --- /dev/null +++ b/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h @@ -0,0 +1,48 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _ALSA_PCM_audio_h +#define _ALSA_PCM_audio_h + +#include + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The audio device handle */ + snd_pcm_t *pcm_handle; + + /* Raw mixing buffer */ + Uint8 *mixbuf; + int mixlen; +}; + +/* Old variable names */ +#define pcm_handle (this->hidden->pcm_handle) +#define mixbuf (this->hidden->mixbuf) +#define mixlen (this->hidden->mixlen) + +#endif /* _ALSA_PCM_audio_h */ diff --git a/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c b/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c new file mode 100644 index 0000000000..373f8c1677 --- /dev/null +++ b/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c @@ -0,0 +1,362 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#ifdef HAVE_SIGNAL_H +#include +#endif +#include + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_artsaudio.h" + +#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC +#include "SDL_name.h" +#include "SDL_loadso.h" +#else +#define SDL_NAME(X) X +#endif + +/* The tag name used by artsc audio */ +#define ARTS_DRIVER_NAME "arts" + +/* Audio driver functions */ +static int ARTS_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void ARTS_WaitAudio(_THIS); +static void ARTS_PlayAudio(_THIS); +static Uint8 *ARTS_GetAudioBuf(_THIS); +static void ARTS_CloseAudio(_THIS); + +#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC + +static const char *arts_library = SDL_AUDIO_DRIVER_ARTS_DYNAMIC; +static void *arts_handle = NULL; +static int arts_loaded = 0; + +static int (*SDL_NAME(arts_init))(void); +static void (*SDL_NAME(arts_free))(void); +static arts_stream_t (*SDL_NAME(arts_play_stream))(int rate, int bits, int channels, const char *name); +static int (*SDL_NAME(arts_stream_set))(arts_stream_t s, arts_parameter_t param, int value); +static int (*SDL_NAME(arts_stream_get))(arts_stream_t s, arts_parameter_t param); +static int (*SDL_NAME(arts_write))(arts_stream_t s, const void *buffer, int count); +static void (*SDL_NAME(arts_close_stream))(arts_stream_t s); +static int (*SDL_NAME(arts_suspend))(void); +static int (*SDL_NAME(arts_suspended))(void); +static const char *(*SDL_NAME(arts_error_text))(int errorcode); + +static struct { + const char *name; + void **func; +} arts_functions[] = { + { "arts_init", (void **)&SDL_NAME(arts_init) }, + { "arts_free", (void **)&SDL_NAME(arts_free) }, + { "arts_play_stream", (void **)&SDL_NAME(arts_play_stream) }, + { "arts_stream_set", (void **)&SDL_NAME(arts_stream_set) }, + { "arts_stream_get", (void **)&SDL_NAME(arts_stream_get) }, + { "arts_write", (void **)&SDL_NAME(arts_write) }, + { "arts_close_stream", (void **)&SDL_NAME(arts_close_stream) }, + { "arts_suspend", (void **)&SDL_NAME(arts_suspend) }, + { "arts_suspended", (void **)&SDL_NAME(arts_suspended) }, + { "arts_error_text", (void **)&SDL_NAME(arts_error_text) }, +}; + +static void UnloadARTSLibrary() +{ + if ( arts_loaded ) { + SDL_UnloadObject(arts_handle); + arts_handle = NULL; + arts_loaded = 0; + } +} + +static int LoadARTSLibrary(void) +{ + int i, retval = -1; + + arts_handle = SDL_LoadObject(arts_library); + if ( arts_handle ) { + arts_loaded = 1; + retval = 0; + for ( i=0; ihidden); + SDL_free(device); + UnloadARTSLibrary(); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + LoadARTSLibrary(); + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + stream = 0; + + /* Set the function pointers */ + this->OpenAudio = ARTS_OpenAudio; + this->WaitAudio = ARTS_WaitAudio; + this->PlayAudio = ARTS_PlayAudio; + this->GetAudioBuf = ARTS_GetAudioBuf; + this->CloseAudio = ARTS_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap ARTS_bootstrap = { + ARTS_DRIVER_NAME, "Analog Realtime Synthesizer", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void ARTS_WaitAudio(_THIS) +{ + Sint32 ticks; + + /* Check to see if the thread-parent process is still alive */ + { static int cnt = 0; + /* Note that this only works with thread implementations + that use a different process id for each thread. + */ + if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ + if ( kill(parent, 0) < 0 ) { + this->enabled = 0; + } + } + } + + /* Use timer for general audio synchronization */ + ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; + if ( ticks > 0 ) { + SDL_Delay(ticks); + } +} + +static void ARTS_PlayAudio(_THIS) +{ + int written; + + /* Write the audio data */ + written = SDL_NAME(arts_write)(stream, mixbuf, mixlen); + + /* If timer synchronization is enabled, set the next write frame */ + if ( frame_ticks ) { + next_frame += frame_ticks; + } + + /* If we couldn't write, assume fatal error for now */ + if ( written < 0 ) { + this->enabled = 0; + } +#ifdef DEBUG_AUDIO + fprintf(stderr, "Wrote %d bytes of audio data\n", written); +#endif +} + +static Uint8 *ARTS_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void ARTS_CloseAudio(_THIS) +{ + if ( mixbuf != NULL ) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if ( stream ) { + SDL_NAME(arts_close_stream)(stream); + stream = 0; + } + SDL_NAME(arts_free)(); +} + +static int ARTS_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + int bits, frag_spec; + Uint16 test_format, format; + int error_code; + + /* Reset the timer synchronization flag */ + frame_ticks = 0.0; + + mixbuf = NULL; + + /* Try for a closest match on audio format */ + format = 0; + bits = 0; + for ( test_format = SDL_FirstAudioFormat(spec->format); + ! format && test_format; ) { +#ifdef DEBUG_AUDIO + fprintf(stderr, "Trying format 0x%4.4x\n", test_format); +#endif + switch ( test_format ) { + case AUDIO_U8: + bits = 8; + format = 1; + break; + case AUDIO_S16LSB: + bits = 16; + format = 1; + break; + default: + format = 0; + break; + } + if ( ! format ) { + test_format = SDL_NextAudioFormat(); + } + } + if ( format == 0 ) { + SDL_SetError("Couldn't find any hardware audio formats"); + return(-1); + } + spec->format = test_format; + + error_code = SDL_NAME(arts_init)(); + if ( error_code != 0 ) { + SDL_SetError("Unable to initialize ARTS: %s", SDL_NAME(arts_error_text)(error_code)); + return(-1); + } + if ( ! ARTS_Suspend() ) { + SDL_SetError("ARTS can not open audio device"); + return(-1); + } + stream = SDL_NAME(arts_play_stream)(spec->freq, bits, spec->channels, "SDL"); + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Determine the power of two of the fragment size */ + for ( frag_spec = 0; (0x01<size; ++frag_spec ); + if ( (0x01<size ) { + SDL_SetError("Fragment size must be a power of two"); + return(-1); + } + frag_spec |= 0x00020000; /* two fragments, for low latency */ + +#ifdef ARTS_P_PACKET_SETTINGS + SDL_NAME(arts_stream_set)(stream, ARTS_P_PACKET_SETTINGS, frag_spec); +#else + SDL_NAME(arts_stream_set)(stream, ARTS_P_PACKET_SIZE, frag_spec&0xffff); + SDL_NAME(arts_stream_set)(stream, ARTS_P_PACKET_COUNT, frag_spec>>16); +#endif + spec->size = SDL_NAME(arts_stream_get)(stream, ARTS_P_PACKET_SIZE); + + /* Allocate mixing buffer */ + mixlen = spec->size; + mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); + if ( mixbuf == NULL ) { + return(-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* Get the parent process id (we're the parent of the audio thread) */ + parent = getpid(); + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h b/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h new file mode 100644 index 0000000000..de3b22822c --- /dev/null +++ b/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h @@ -0,0 +1,60 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_artscaudio_h +#define _SDL_artscaudio_h + +#include + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The stream descriptor for the audio device */ + arts_stream_t stream; + + /* The parent process id, to detect when application quits */ + pid_t parent; + + /* Raw mixing buffer */ + Uint8 *mixbuf; + int mixlen; + + /* Support for audio timing using a timer, in addition to select() */ + float frame_ticks; + float next_frame; +}; +#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */ + +/* Old variable names */ +#define stream (this->hidden->stream) +#define parent (this->hidden->parent) +#define mixbuf (this->hidden->mixbuf) +#define mixlen (this->hidden->mixlen) +#define frame_ticks (this->hidden->frame_ticks) +#define next_frame (this->hidden->next_frame) + +#endif /* _SDL_artscaudio_h */ + diff --git a/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc b/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc new file mode 100644 index 0000000000..de635f8bad --- /dev/null +++ b/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc @@ -0,0 +1,225 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to the audio stream on BeOS */ + +#include + +#include "../../main/beos/SDL_BeApp.h" + +extern "C" { + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" +#include "../../thread/beos/SDL_systhread_c.h" +#include "SDL_beaudio.h" + + +/* Audio driver functions */ +static int BE_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void BE_WaitAudio(_THIS); +static void BE_PlayAudio(_THIS); +static Uint8 *BE_GetAudioBuf(_THIS); +static void BE_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *device; + + /* Initialize all variables that we clean on shutdown */ + device = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( device ) { + SDL_memset(device, 0, (sizeof *device)); + device->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *device->hidden)); + } + if ( (device == NULL) || (device->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( device ) { + SDL_free(device); + } + return(0); + } + SDL_memset(device->hidden, 0, (sizeof *device->hidden)); + + /* Set the function pointers */ + device->OpenAudio = BE_OpenAudio; + device->WaitAudio = BE_WaitAudio; + device->PlayAudio = BE_PlayAudio; + device->GetAudioBuf = BE_GetAudioBuf; + device->CloseAudio = BE_CloseAudio; + + device->free = Audio_DeleteDevice; + + return device; +} + +AudioBootStrap BAUDIO_bootstrap = { + "baudio", "BeOS BSoundPlayer", + Audio_Available, Audio_CreateDevice +}; + +/* The BeOS callback for handling the audio buffer */ +static void FillSound(void *device, void *stream, size_t len, + const media_raw_audio_format &format) +{ + SDL_AudioDevice *audio = (SDL_AudioDevice *)device; + + /* Silence the buffer, since it's ours */ + SDL_memset(stream, audio->spec.silence, len); + + /* Only do soemthing if audio is enabled */ + if ( ! audio->enabled ) + return; + + if ( ! audio->paused ) { + if ( audio->convert.needed ) { + SDL_mutexP(audio->mixer_lock); + (*audio->spec.callback)(audio->spec.userdata, + (Uint8 *)audio->convert.buf,audio->convert.len); + SDL_mutexV(audio->mixer_lock); + SDL_ConvertAudio(&audio->convert); + SDL_memcpy(stream,audio->convert.buf,audio->convert.len_cvt); + } else { + SDL_mutexP(audio->mixer_lock); + (*audio->spec.callback)(audio->spec.userdata, + (Uint8 *)stream, len); + SDL_mutexV(audio->mixer_lock); + } + } + return; +} + +/* Dummy functions -- we don't use thread-based audio */ +void BE_WaitAudio(_THIS) +{ + return; +} +void BE_PlayAudio(_THIS) +{ + return; +} +Uint8 *BE_GetAudioBuf(_THIS) +{ + return(NULL); +} + +void BE_CloseAudio(_THIS) +{ + if ( audio_obj ) { + audio_obj->Stop(); + delete audio_obj; + audio_obj = NULL; + } + + /* Quit the Be Application, if there's nothing left to do */ + SDL_QuitBeApp(); +} + +int BE_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + int valid_datatype = 0; + media_raw_audio_format format; + Uint16 test_format = SDL_FirstAudioFormat(spec->format); + + /* Parse the audio format and fill the Be raw audio format */ + memset(&format, '\0', sizeof (media_raw_audio_format)); + format.byte_order = B_MEDIA_LITTLE_ENDIAN; + format.frame_rate = (float) spec->freq; + format.channel_count = spec->channels; /* !!! FIXME: support > 2? */ + while ((!valid_datatype) && (test_format)) { + valid_datatype = 1; + spec->format = test_format; + switch (test_format) { + case AUDIO_S8: + format.format = media_raw_audio_format::B_AUDIO_CHAR; + break; + + case AUDIO_U8: + format.format = media_raw_audio_format::B_AUDIO_UCHAR; + break; + + case AUDIO_S16LSB: + format.format = media_raw_audio_format::B_AUDIO_SHORT; + break; + + case AUDIO_S16MSB: + format.format = media_raw_audio_format::B_AUDIO_SHORT; + format.byte_order = B_MEDIA_BIG_ENDIAN; + break; + + default: + valid_datatype = 0; + test_format = SDL_NextAudioFormat(); + break; + } + } + + if (!valid_datatype) { /* shouldn't happen, but just in case... */ + SDL_SetError("Unsupported audio format"); + return (-1); + } + + /* Initialize the Be Application, if it's not already started */ + if (SDL_InitBeApp() < 0) { + return (-1); + } + + format.buffer_size = spec->samples; + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Subscribe to the audio stream (creates a new thread) */ + { sigset_t omask; + SDL_MaskSignals(&omask); + audio_obj = new BSoundPlayer(&format, "SDL Audio", FillSound, + NULL, _this); + SDL_UnmaskSignals(&omask); + } + if ( audio_obj->Start() == B_NO_ERROR ) { + audio_obj->SetHasData(true); + } else { + SDL_SetError("Unable to start Be audio"); + return(-1); + } + + /* We're running! */ + return(1); +} + +}; /* Extern C */ diff --git a/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h b/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h new file mode 100644 index 0000000000..adaf1dee5d --- /dev/null +++ b/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h @@ -0,0 +1,39 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_lowaudio_h +#define _SDL_lowaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *_this + +struct SDL_PrivateAudioData { + BSoundPlayer *audio_obj; +}; + +/* Old variable names */ +#define audio_obj (_this->hidden->audio_obj) + +#endif /* _SDL_lowaudio_h */ diff --git a/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c b/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c new file mode 100644 index 0000000000..e5e0d9480a --- /dev/null +++ b/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c @@ -0,0 +1,404 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + * Driver for native OpenBSD/NetBSD audio(4). + * vedge@vedge.com.ar. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_bsdaudio.h" + +/* The tag name used by NetBSD/OpenBSD audio */ +#ifdef __NetBSD__ +#define BSD_AUDIO_DRIVER_NAME "netbsd" +#define BSD_AUDIO_DRIVER_DESC "Native NetBSD audio" +#else +#define BSD_AUDIO_DRIVER_NAME "openbsd" +#define BSD_AUDIO_DRIVER_DESC "Native OpenBSD audio" +#endif + +/* Open the audio device for playback, and don't block if busy */ +/* #define USE_BLOCKING_WRITES */ + +/* Use timer for synchronization */ +/* #define USE_TIMER_SYNC */ + +/* #define DEBUG_AUDIO */ +/* #define DEBUG_AUDIO_STREAM */ + +#ifdef USE_BLOCKING_WRITES +#define OPEN_FLAGS O_WRONLY +#else +#define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) +#endif + +/* Audio driver functions */ +static void OBSD_WaitAudio(_THIS); +static int OBSD_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void OBSD_PlayAudio(_THIS); +static Uint8 *OBSD_GetAudioBuf(_THIS); +static void OBSD_CloseAudio(_THIS); + +#ifdef DEBUG_AUDIO +static void OBSD_Status(_THIS); +#endif + +/* Audio driver bootstrap functions */ + +static int +Audio_Available(void) +{ + int fd; + int available; + + available = 0; + fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0); + if(fd >= 0) { + available = 1; + close(fd); + } + return(available); +} + +static void +Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice +*Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice*)SDL_malloc(sizeof(SDL_AudioDevice)); + if(this) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = + (struct SDL_PrivateAudioData*)SDL_malloc((sizeof *this->hidden)); + } + if((this == NULL) || (this->hidden == NULL)) { + SDL_OutOfMemory(); + if(this) SDL_free(this); + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + audio_fd = -1; + + /* Set the function pointers */ + this->OpenAudio = OBSD_OpenAudio; + this->WaitAudio = OBSD_WaitAudio; + this->PlayAudio = OBSD_PlayAudio; + this->GetAudioBuf = OBSD_GetAudioBuf; + this->CloseAudio = OBSD_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap BSD_AUDIO_bootstrap = { + BSD_AUDIO_DRIVER_NAME, BSD_AUDIO_DRIVER_DESC, + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void +OBSD_WaitAudio(_THIS) +{ +#ifndef USE_BLOCKING_WRITES /* Not necessary when using blocking writes */ + /* See if we need to use timed audio synchronization */ + if ( frame_ticks ) { + /* Use timer for general audio synchronization */ + Sint32 ticks; + + ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; + if ( ticks > 0 ) { + SDL_Delay(ticks); + } + } else { + /* Use select() for audio synchronization */ + fd_set fdset; + struct timeval timeout; + + FD_ZERO(&fdset); + FD_SET(audio_fd, &fdset); + timeout.tv_sec = 10; + timeout.tv_usec = 0; +#ifdef DEBUG_AUDIO + fprintf(stderr, "Waiting for audio to get ready\n"); +#endif + if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 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); + this->enabled = 0; + /* Don't try to close - may hang */ + 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 +OBSD_PlayAudio(_THIS) +{ + int written, p=0; + + /* Write the audio data, checking for EAGAIN on broken audio drivers */ + do { + written = write(audio_fd, &mixbuf[p], 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; + } + + if ( p < written || ((written < 0) && ((errno == 0) || (errno == EAGAIN))) ) { + SDL_Delay(1); /* Let a little CPU time go by */ + } + } while ( p < written ); + + /* If timer synchronization is enabled, set the next write frame */ + if ( frame_ticks ) { + next_frame += frame_ticks; + } + + /* If we couldn't write, assume fatal error for now */ + if ( written < 0 ) { + this->enabled = 0; + } +#ifdef DEBUG_AUDIO + fprintf(stderr, "Wrote %d bytes of audio data\n", written); +#endif +} + +static Uint8 +*OBSD_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void +OBSD_CloseAudio(_THIS) +{ + if(mixbuf != NULL) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if(audio_fd >= 0) { + close(audio_fd); + audio_fd = -1; + } +} + +#ifdef DEBUG_AUDIO +void +OBSD_Status(_THIS) +{ + audio_info_t info; + + if(ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) { + fprintf(stderr,"AUDIO_GETINFO failed.\n"); + return; + } + + fprintf(stderr,"\n" +"[play/record info]\n" +"buffer size : %d bytes\n" +"sample rate : %i Hz\n" +"channels : %i\n" +"precision : %i-bit\n" +"encoding : 0x%x\n" +"seek : %i\n" +"sample count : %i\n" +"EOF count : %i\n" +"paused : %s\n" +"error occured : %s\n" +"waiting : %s\n" +"active : %s\n" +"", + info.play.buffer_size, + info.play.sample_rate, + info.play.channels, + info.play.precision, + info.play.encoding, + info.play.seek, + info.play.samples, + info.play.eof, + info.play.pause ? "yes" : "no", + info.play.error ? "yes" : "no", + info.play.waiting ? "yes" : "no", + info.play.active ? "yes": "no"); + + fprintf(stderr,"\n" +"[audio info]\n" +"monitor_gain : %i\n" +"hw block size : %d bytes\n" +"hi watermark : %i\n" +"lo watermark : %i\n" +"audio mode : %s\n" +"", + info.monitor_gain, + info.blocksize, + info.hiwat, info.lowat, + (info.mode == AUMODE_PLAY) ? "PLAY" + : (info.mode = AUMODE_RECORD) ? "RECORD" + : (info.mode == AUMODE_PLAY_ALL ? "PLAY_ALL" + : "?")); +} +#endif /* DEBUG_AUDIO */ + +static int +OBSD_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + char audiodev[64]; + Uint16 format; + audio_info_t info; + + AUDIO_INITINFO(&info); + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + +#ifdef USE_TIMER_SYNC + frame_ticks = 0.0; +#endif + + /* Open the audio device */ + audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0); + if(audio_fd < 0) { + SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); + return(-1); + } + + /* Set to play mode */ + info.mode = AUMODE_PLAY; + if(ioctl(audio_fd, AUDIO_SETINFO, &info) < 0) { + SDL_SetError("Couldn't put device into play mode"); + return(-1); + } + + mixbuf = NULL; + AUDIO_INITINFO(&info); + for (format = SDL_FirstAudioFormat(spec->format); + format; format = SDL_NextAudioFormat()) + { + switch(format) { + case AUDIO_U8: + info.play.encoding = AUDIO_ENCODING_ULINEAR; + info.play.precision = 8; + break; + case AUDIO_S8: + info.play.encoding = AUDIO_ENCODING_SLINEAR; + info.play.precision = 8; + break; + case AUDIO_S16LSB: + info.play.encoding = AUDIO_ENCODING_SLINEAR_LE; + info.play.precision = 16; + break; + case AUDIO_S16MSB: + info.play.encoding = AUDIO_ENCODING_SLINEAR_BE; + info.play.precision = 16; + break; + case AUDIO_U16LSB: + info.play.encoding = AUDIO_ENCODING_ULINEAR_LE; + info.play.precision = 16; + break; + case AUDIO_U16MSB: + info.play.encoding = AUDIO_ENCODING_ULINEAR_BE; + info.play.precision = 16; + break; + default: + continue; + } + if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0) + break; + } + + if(!format) { + SDL_SetError("No supported encoding for 0x%x", spec->format); + return(-1); + } + + spec->format = format; + + AUDIO_INITINFO(&info); + info.play.channels = spec->channels; + if (ioctl(audio_fd, AUDIO_SETINFO, &info) == -1) + spec->channels = 1; + AUDIO_INITINFO(&info); + info.play.sample_rate = spec->freq; + info.blocksize = spec->size; + info.hiwat = 5; + info.lowat = 3; + (void)ioctl(audio_fd, AUDIO_SETINFO, &info); + (void)ioctl(audio_fd, AUDIO_GETINFO, &info); + spec->freq = info.play.sample_rate; + /* Allocate mixing buffer */ + mixlen = spec->size; + mixbuf = (Uint8*)SDL_AllocAudioMem(mixlen); + if(mixbuf == NULL) { + return(-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* Get the parent process id (we're the parent of the audio thread) */ + parent = getpid(); + +#ifdef DEBUG_AUDIO + OBSD_Status(this); +#endif + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h b/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h new file mode 100644 index 0000000000..c9f69cf544 --- /dev/null +++ b/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h @@ -0,0 +1,58 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_openbsdaudio_h +#define _SDL_openbsdaudio_h + +#include "../SDL_sysaudio.h" + +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData +{ + /* The file descriptor for the audio device */ + int audio_fd; + + /* The parent process id, to detect when application quits */ + pid_t parent; + + /* Raw mixing buffer */ + Uint8 *mixbuf; + int mixlen; + + /* Support for audio timing using a timer, in addition to select() */ + float frame_ticks; + float next_frame; +}; + +#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */ + +/* Old variable names */ +#define audio_fd (this->hidden->audio_fd) +#define parent (this->hidden->parent) +#define mixbuf (this->hidden->mixbuf) +#define mixlen (this->hidden->mixlen) +#define frame_ticks (this->hidden->frame_ticks) +#define next_frame (this->hidden->next_frame) + +#endif /* _SDL_openbsdaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dart/SDL_dart.c b/apps/plugins/sdl/src/audio/dart/SDL_dart.c new file mode 100644 index 0000000000..77e530db51 --- /dev/null +++ b/apps/plugins/sdl/src/audio/dart/SDL_dart.c @@ -0,0 +1,441 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "SDL_dart.h" + +// Buffer states: +#define BUFFER_EMPTY 0 +#define BUFFER_USED 1 + +typedef struct _tMixBufferDesc { + int iBufferUsage; // BUFFER_EMPTY or BUFFER_USED + SDL_AudioDevice *pSDLAudioDevice; +} tMixBufferDesc, *pMixBufferDesc; + + +//--------------------------------------------------------------------- +// DARTEventFunc +// +// This function is called by DART, when an event occures, like end of +// playback of a buffer, etc... +//--------------------------------------------------------------------- +LONG APIENTRY DARTEventFunc(ULONG ulStatus, + PMCI_MIX_BUFFER pBuffer, + ULONG ulFlags) +{ + if (ulFlags && MIX_WRITE_COMPLETE) + { // Playback of buffer completed! + + // Get pointer to buffer description + pMixBufferDesc pBufDesc; + + if (pBuffer) + { + pBufDesc = (pMixBufferDesc) (*pBuffer).ulUserParm; + + if (pBufDesc) + { + SDL_AudioDevice *pSDLAudioDevice = pBufDesc->pSDLAudioDevice; + // Set the buffer to be empty + pBufDesc->iBufferUsage = BUFFER_EMPTY; + // And notify DART feeder thread that it will have to work a bit. + if (pSDLAudioDevice) + DosPostEventSem(pSDLAudioDevice->hidden->hevAudioBufferPlayed); + } + } + } + return TRUE; +} + + +int DART_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + Uint16 test_format = SDL_FirstAudioFormat(spec->format); + int valid_datatype = 0; + MCI_AMP_OPEN_PARMS AmpOpenParms; + MCI_GENERIC_PARMS GenericParms; + int iDeviceOrd = 0; // Default device to be used + int bOpenShared = 1; // Try opening it shared + int iBits = 16; // Default is 16 bits signed + int iFreq = 44100; // Default is 44KHz + int iChannels = 2; // Default is 2 channels (Stereo) + int iNumBufs = 2; // Number of audio buffers: 2 + int iBufSize; + int iOpenMode; + int iSilence; + int rc; + + // First thing is to try to open a given DART device! + SDL_memset(&AmpOpenParms, 0, sizeof(MCI_AMP_OPEN_PARMS)); + // pszDeviceType should contain the device type in low word, and device ordinal in high word! + AmpOpenParms.pszDeviceType = (PSZ) (MCI_DEVTYPE_AUDIO_AMPMIX | (iDeviceOrd << 16)); + + iOpenMode = MCI_WAIT | MCI_OPEN_TYPE_ID; + if (bOpenShared) iOpenMode |= MCI_OPEN_SHAREABLE; + + rc = mciSendCommand( 0, MCI_OPEN, + iOpenMode, + (PVOID) &AmpOpenParms, 0); + if (rc!=MCIERR_SUCCESS) // No audio available?? + return (-1); + // Save the device ID we got from DART! + // We will use this in the next calls! + iDeviceOrd = AmpOpenParms.usDeviceID; + + // Determine the audio parameters from the AudioSpec + if (spec->channels > 2) + spec->channels = 2; // !!! FIXME: more than stereo support in OS/2? + + while ((!valid_datatype) && (test_format)) { + spec->format = test_format; + valid_datatype = 1; + switch (test_format) { + case AUDIO_U8: + // Unsigned 8 bit audio data + iSilence = 0x80; + iBits = 8; + break; + + case AUDIO_S16LSB: + // Signed 16 bit audio data + iSilence = 0x00; + iBits = 16; + break; + + default: + valid_datatype = 0; + test_format = SDL_NextAudioFormat(); + break; + } + } + + if (!valid_datatype) { // shouldn't happen, but just in case... + // Close DART, and exit with error code! + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("Unsupported audio format"); + return (-1); + } + + iFreq = spec->freq; + iChannels = spec->channels; + /* Update the fragment size as size in bytes */ + SDL_CalculateAudioSpec(spec); + iBufSize = spec->size; + + // Now query this device if it supports the given freq/bits/channels! + SDL_memset(&(_this->hidden->MixSetupParms), 0, sizeof(MCI_MIXSETUP_PARMS)); + _this->hidden->MixSetupParms.ulBitsPerSample = iBits; + _this->hidden->MixSetupParms.ulFormatTag = MCI_WAVE_FORMAT_PCM; + _this->hidden->MixSetupParms.ulSamplesPerSec = iFreq; + _this->hidden->MixSetupParms.ulChannels = iChannels; + _this->hidden->MixSetupParms.ulFormatMode = MCI_PLAY; + _this->hidden->MixSetupParms.ulDeviceType = MCI_DEVTYPE_WAVEFORM_AUDIO; + _this->hidden->MixSetupParms.pmixEvent = DARTEventFunc; + rc = mciSendCommand (iDeviceOrd, MCI_MIXSETUP, + MCI_WAIT | MCI_MIXSETUP_QUERYMODE, + &(_this->hidden->MixSetupParms), 0); + if (rc!=MCIERR_SUCCESS) + { // The device cannot handle this format! + // Close DART, and exit with error code! + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("Audio device doesn't support requested audio format"); + return(-1); + } + // The device can handle this format, so initialize! + rc = mciSendCommand(iDeviceOrd, MCI_MIXSETUP, + MCI_WAIT | MCI_MIXSETUP_INIT, + &(_this->hidden->MixSetupParms), 0); + if (rc!=MCIERR_SUCCESS) + { // The device could not be opened! + // Close DART, and exit with error code! + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("Audio device could not be set up"); + return(-1); + } + // Ok, the device is initialized. + // Now we should allocate buffers. For this, we need a place where + // the buffer descriptors will be: + _this->hidden->pMixBuffers = (MCI_MIX_BUFFER *) SDL_malloc(sizeof(MCI_MIX_BUFFER)*iNumBufs); + if (!(_this->hidden->pMixBuffers)) + { // Not enough memory! + // Close DART, and exit with error code! + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("Not enough memory for audio buffer descriptors"); + return(-1); + } + // Now that we have the place for buffer list, we can ask DART for the + // buffers! + _this->hidden->BufferParms.ulNumBuffers = iNumBufs; // Number of buffers + _this->hidden->BufferParms.ulBufferSize = iBufSize; // each with this size + _this->hidden->BufferParms.pBufList = _this->hidden->pMixBuffers; // getting descriptorts into this list + // Allocate buffers! + rc = mciSendCommand(iDeviceOrd, MCI_BUFFER, + MCI_WAIT | MCI_ALLOCATE_MEMORY, + &(_this->hidden->BufferParms), 0); + if ((rc!=MCIERR_SUCCESS) || (iNumBufs != _this->hidden->BufferParms.ulNumBuffers) || (_this->hidden->BufferParms.ulBufferSize==0)) + { // Could not allocate memory! + // Close DART, and exit with error code! + SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL; + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("DART could not allocate buffers"); + return(-1); + } + // Ok, we have all the buffers allocated, let's mark them! + { + int i; + for (i=0; ihidden->pMixBuffers[i].pBuffer)) || (!pBufferDesc)) + { // Wrong buffer! + // Close DART, and exit with error code! + // Free buffer descriptions + { int j; + for (j=0; jhidden->pMixBuffers[j].ulUserParm)); + } + // and cleanup + mciSendCommand(iDeviceOrd, MCI_BUFFER, MCI_WAIT | MCI_DEALLOCATE_MEMORY, &(_this->hidden->BufferParms), 0); + SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL; + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("Error at internal buffer check"); + return(-1); + } + pBufferDesc->iBufferUsage = BUFFER_EMPTY; + pBufferDesc->pSDLAudioDevice = _this; + + _this->hidden->pMixBuffers[i].ulBufferLength = _this->hidden->BufferParms.ulBufferSize; + _this->hidden->pMixBuffers[i].ulUserParm = (ULONG) pBufferDesc; // User parameter: Description of buffer + _this->hidden->pMixBuffers[i].ulFlags = 0; // Some stuff should be flagged here for DART, like end of + // audio data, but as we will continously send + // audio data, there will be no end.:) + SDL_memset(_this->hidden->pMixBuffers[i].pBuffer, iSilence, iBufSize); + } + } + _this->hidden->iNextFreeBuffer = 0; + _this->hidden->iLastPlayedBuf = -1; + // Create event semaphore + if (DosCreateEventSem(NULL, &(_this->hidden->hevAudioBufferPlayed), 0, FALSE)!=NO_ERROR) + { + // Could not create event semaphore! + { + int i; + for (i=0; ihidden->pMixBuffers[i].ulUserParm)); + } + mciSendCommand(iDeviceOrd, MCI_BUFFER, MCI_WAIT | MCI_DEALLOCATE_MEMORY, &(_this->hidden->BufferParms), 0); + SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL; + mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0); + SDL_SetError("Could not create event semaphore"); + return(-1); + } + + // Store the new settings in global variables + _this->hidden->iCurrDeviceOrd = iDeviceOrd; + _this->hidden->iCurrFreq = iFreq; + _this->hidden->iCurrBits = iBits; + _this->hidden->iCurrChannels = iChannels; + _this->hidden->iCurrNumBufs = iNumBufs; + _this->hidden->iCurrBufSize = iBufSize; + + return (0); +} + + + +void DART_ThreadInit(_THIS) +{ + return; +} + +/* This function waits until it is possible to write a full sound buffer */ +void DART_WaitAudio(_THIS) +{ + int i; + pMixBufferDesc pBufDesc; + ULONG ulPostCount; + + DosResetEventSem(_this->hidden->hevAudioBufferPlayed, &ulPostCount); + // If there is already an empty buffer, then return now! + for (i=0; i<_this->hidden->iCurrNumBufs; i++) + { + pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[i].ulUserParm; + if (pBufDesc->iBufferUsage == BUFFER_EMPTY) + return; + } + // If there is no empty buffer, wait for one to be empty! + DosWaitEventSem(_this->hidden->hevAudioBufferPlayed, 1000); // Wait max 1 sec!!! Important! + return; +} + +void DART_PlayAudio(_THIS) +{ + int iFreeBuf = _this->hidden->iNextFreeBuffer; + pMixBufferDesc pBufDesc; + + pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[iFreeBuf].ulUserParm; + pBufDesc->iBufferUsage = BUFFER_USED; + // Send it to DART to be queued + _this->hidden->MixSetupParms.pmixWrite(_this->hidden->MixSetupParms.ulMixHandle, + &(_this->hidden->pMixBuffers[iFreeBuf]), 1); + + _this->hidden->iLastPlayedBuf = iFreeBuf; + iFreeBuf = (iFreeBuf+1) % _this->hidden->iCurrNumBufs; + _this->hidden->iNextFreeBuffer = iFreeBuf; +} + +Uint8 *DART_GetAudioBuf(_THIS) +{ + int iFreeBuf; + Uint8 *pResult; + pMixBufferDesc pBufDesc; + + if (_this) + { + if (_this->hidden) + { + iFreeBuf = _this->hidden->iNextFreeBuffer; + pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[iFreeBuf].ulUserParm; + + if (pBufDesc) + { + if (pBufDesc->iBufferUsage == BUFFER_EMPTY) + { + pResult = _this->hidden->pMixBuffers[iFreeBuf].pBuffer; + return pResult; + } + } else + printf("[DART_GetAudioBuf] : ERROR! pBufDesc = %p\n", pBufDesc); + } else + printf("[DART_GetAudioBuf] : ERROR! _this->hidden = %p\n", _this->hidden); + } else + printf("[DART_GetAudioBuf] : ERROR! _this = %p\n", _this); + return NULL; +} + +void DART_WaitDone(_THIS) +{ + pMixBufferDesc pBufDesc; + ULONG ulPostCount; + APIRET rc; + + pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[_this->hidden->iLastPlayedBuf].ulUserParm; + rc = NO_ERROR; + while ((pBufDesc->iBufferUsage != BUFFER_EMPTY) && (rc==NO_ERROR)) + { + DosResetEventSem(_this->hidden->hevAudioBufferPlayed, &ulPostCount); + rc = DosWaitEventSem(_this->hidden->hevAudioBufferPlayed, 1000); // 1 sec timeout! Important! + } +} + +void DART_CloseAudio(_THIS) +{ + MCI_GENERIC_PARMS GenericParms; + int rc; + + // Stop DART playback + rc = mciSendCommand(_this->hidden->iCurrDeviceOrd, MCI_STOP, MCI_WAIT, &GenericParms, 0); + if (rc!=MCIERR_SUCCESS) + { +#ifdef SFX_DEBUG_BUILD + printf("Could not stop DART playback!\n"); + fflush(stdout); +#endif + } + + // Close event semaphore + DosCloseEventSem(_this->hidden->hevAudioBufferPlayed); + + // Free memory of buffer descriptions + { + int i; + for (i=0; i<_this->hidden->iCurrNumBufs; i++) SDL_free((void *)(_this->hidden->pMixBuffers[i].ulUserParm)); + } + + // Deallocate buffers + rc = mciSendCommand(_this->hidden->iCurrDeviceOrd, MCI_BUFFER, MCI_WAIT | MCI_DEALLOCATE_MEMORY, &(_this->hidden->BufferParms), 0); + + // Free bufferlist + SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL; + + // Close dart + rc = mciSendCommand(_this->hidden->iCurrDeviceOrd, MCI_CLOSE, MCI_WAIT, &(GenericParms), 0); +} + +/* Audio driver bootstrap functions */ + +int Audio_Available(void) +{ + return(1); +} + +void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) + { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) + { + SDL_OutOfMemory(); + if ( this ) + SDL_free(this); + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = DART_OpenAudio; + this->ThreadInit = DART_ThreadInit; + this->WaitAudio = DART_WaitAudio; + this->PlayAudio = DART_PlayAudio; + this->GetAudioBuf = DART_GetAudioBuf; + this->WaitDone = DART_WaitDone; + this->CloseAudio = DART_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap DART_bootstrap = { + "dart", "OS/2 Direct Audio RouTines (DART)", + Audio_Available, Audio_CreateDevice +}; + diff --git a/apps/plugins/sdl/src/audio/dart/SDL_dart.h b/apps/plugins/sdl/src/audio/dart/SDL_dart.h new file mode 100644 index 0000000000..68c27bd9d3 --- /dev/null +++ b/apps/plugins/sdl/src/audio/dart/SDL_dart.h @@ -0,0 +1,63 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_lowaudio_h +#define _SDL_lowaudio_h + +#define INCL_TYPES +#define INCL_DOSSEMAPHORES +#define INCL_DOSRESOURCES +#define INCL_DOSMISC +#define INCL_DOSERRORS + +#define INCL_OS2MM +#define INCL_MMIOOS2 +#define INCL_MCIOS2 +#include +#include // DART stuff and MMIO stuff + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the audio functions */ +#define _THIS SDL_AudioDevice *_this + +/* The DirectSound objects */ +struct SDL_PrivateAudioData +{ + int iCurrDeviceOrd; + int iCurrFreq; + int iCurrBits; + int iCurrChannels; + int iCurrNumBufs; + int iCurrBufSize; + + int iLastPlayedBuf; + int iNextFreeBuffer; + + MCI_BUFFER_PARMS BufferParms; // Sound buffer parameters + MCI_MIX_BUFFER *pMixBuffers; // Sound buffers + MCI_MIXSETUP_PARMS MixSetupParms; // Mixer setup parameters + HEV hevAudioBufferPlayed; // Event semaphore to indicate that an audio buffer has been played by DART +}; + +#endif /* _SDL_lowaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c b/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c new file mode 100644 index 0000000000..88daa8723a --- /dev/null +++ b/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c @@ -0,0 +1,246 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org + +*/ +#include "SDL_config.h" + +/* Output dreamcast aica */ + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_dcaudio.h" + +#include "aica.h" +#include + +/* Audio driver functions */ +static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DCAUD_WaitAudio(_THIS); +static void DCAUD_PlayAudio(_THIS); +static Uint8 *DCAUD_GetAudioBuf(_THIS); +static void DCAUD_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ +static int DCAUD_Available(void) +{ + return 1; +} + +static void DCAUD_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *DCAUD_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = DCAUD_OpenAudio; + this->WaitAudio = DCAUD_WaitAudio; + this->PlayAudio = DCAUD_PlayAudio; + this->GetAudioBuf = DCAUD_GetAudioBuf; + this->CloseAudio = DCAUD_CloseAudio; + + this->free = DCAUD_DeleteDevice; + + spu_init(); + + return this; +} + +AudioBootStrap DCAUD_bootstrap = { + "dcaudio", "Dreamcast AICA audio", + DCAUD_Available, DCAUD_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DCAUD_WaitAudio(_THIS) +{ + if (this->hidden->playing) { + /* wait */ + while(aica_get_pos(0)/this->spec.samples == this->hidden->nextbuf) { + thd_pass(); + } + } +} + +#define SPU_RAM_BASE 0xa0800000 + +static void spu_memload_stereo8(int leftpos,int rightpos,void *src0,size_t size) +{ + uint8 *src = src0; + uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE); + uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE); + size = (size+7)/8; + while(size--) { + unsigned lval,rval; + lval = *src++; + rval = *src++; + lval|= (*src++)<<8; + rval|= (*src++)<<8; + lval|= (*src++)<<16; + rval|= (*src++)<<16; + lval|= (*src++)<<24; + rval|= (*src++)<<24; + g2_write_32(left++,lval); + g2_write_32(right++,rval); + g2_fifo_wait(); + } +} + +static void spu_memload_stereo16(int leftpos,int rightpos,void *src0,size_t size) +{ + uint16 *src = src0; + uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE); + uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE); + size = (size+7)/8; + while(size--) { + unsigned lval,rval; + lval = *src++; + rval = *src++; + lval|= (*src++)<<16; + rval|= (*src++)<<16; + g2_write_32(left++,lval); + g2_write_32(right++,rval); + g2_fifo_wait(); + } +} + +static void DCAUD_PlayAudio(_THIS) +{ + SDL_AudioSpec *spec = &this->spec; + unsigned int offset; + + if (this->hidden->playing) { + /* wait */ + while(aica_get_pos(0)/spec->samples == this->hidden->nextbuf) { + thd_pass(); + } + } + + offset = this->hidden->nextbuf*spec->size; + this->hidden->nextbuf^=1; + /* Write the audio data, checking for EAGAIN on broken audio drivers */ + if (spec->channels==1) { + spu_memload(this->hidden->leftpos+offset,this->hidden->mixbuf,this->hidden->mixlen); + } else { + offset/=2; + if ((this->spec.format&255)==8) { + spu_memload_stereo8(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen); + } else { + spu_memload_stereo16(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen); + } + } + + if (!this->hidden->playing) { + int mode; + this->hidden->playing = 1; + mode = (spec->format==AUDIO_S8)?SM_8BIT:SM_16BIT; + if (spec->channels==1) { + aica_play(0,mode,this->hidden->leftpos,0,spec->samples*2,spec->freq,255,128,1); + } else { + aica_play(0,mode,this->hidden->leftpos ,0,spec->samples*2,spec->freq,255,0,1); + aica_play(1,mode,this->hidden->rightpos,0,spec->samples*2,spec->freq,255,255,1); + } + } +} + +static Uint8 *DCAUD_GetAudioBuf(_THIS) +{ + return(this->hidden->mixbuf); +} + +static void DCAUD_CloseAudio(_THIS) +{ + aica_stop(0); + if (this->spec.channels==2) aica_stop(1); + if ( this->hidden->mixbuf != NULL ) { + SDL_FreeAudioMem(this->hidden->mixbuf); + this->hidden->mixbuf = NULL; + } +} + +static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + Uint16 test_format = SDL_FirstAudioFormat(spec->format); + int valid_datatype = 0; + while ((!valid_datatype) && (test_format)) { + spec->format = test_format; + switch (test_format) { + /* only formats Dreamcast accepts... */ + case AUDIO_S8: + case AUDIO_S16LSB: + valid_datatype = 1; + break; + + default: + test_format = SDL_NextAudioFormat(); + break; + } + } + + if (!valid_datatype) { /* shouldn't happen, but just in case... */ + SDL_SetError("Unsupported audio format"); + return (-1); + } + + if (spec->channels > 2) + spec->channels = 2; /* no more than stereo on the Dreamcast. */ + + /* Update the fragment size as size in bytes */ + SDL_CalculateAudioSpec(spec); + + /* Allocate mixing buffer */ + this->hidden->mixlen = spec->size; + this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); + if ( this->hidden->mixbuf == NULL ) { + return(-1); + } + SDL_memset(this->hidden->mixbuf, spec->silence, spec->size); + this->hidden->leftpos = 0x11000; + this->hidden->rightpos = 0x11000+spec->size; + this->hidden->playing = 0; + this->hidden->nextbuf = 0; + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h b/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h new file mode 100644 index 0000000000..fba95b3eda --- /dev/null +++ b/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h @@ -0,0 +1,41 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_dcaudio_h +#define _SDL_dcaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + Uint8 *mixbuf; + Uint32 mixlen; + int playing; + int leftpos,rightpos; + int nextbuf; +}; + +#endif /* _SDL_dcaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dc/aica.c b/apps/plugins/sdl/src/audio/dc/aica.c new file mode 100644 index 0000000000..b6a1c93644 --- /dev/null +++ b/apps/plugins/sdl/src/audio/dc/aica.c @@ -0,0 +1,271 @@ +/* This file is part of the Dreamcast function library. + * Please see libdream.c for further details. + * + * (c)2000 Dan Potter + * modify BERO + */ +#include "aica.h" + +#include +#include + +/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */ +#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */ + +/* Some convienence macros */ +#define SNDREGADDR(x) (0xa0700000 + (x)) +#define CHNREGADDR(ch,x) SNDREGADDR(0x80*(ch)+(x)) + + +#define SNDREG32(x) (*(volatile unsigned long *)SNDREGADDR(x)) +#define SNDREG8(x) (*(volatile unsigned char *)SNDREGADDR(x)) +#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x)) +#define CHNREG8(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x)) + +#define G2_LOCK(OLD) \ + do { \ + if (!irq_inside_int()) \ + OLD = irq_disable(); \ + /* suspend any G2 DMA here... */ \ + while((*(volatile unsigned int *)0xa05f688c) & 0x20) \ + ; \ + } while(0) + +#define G2_UNLOCK(OLD) \ + do { \ + /* resume any G2 DMA here... */ \ + if (!irq_inside_int()) \ + irq_restore(OLD); \ + } while(0) + + +void aica_init() { + int i, j, old = 0; + + /* Initialize AICA channels */ + G2_LOCK(old); + SNDREG32(0x2800) = 0x0000; + + for (i=0; i<64; i++) { + for (j=0; j<0x80; j+=4) { + if ((j&31)==0) g2_fifo_wait(); + CHNREG32(i, j) = 0; + } + g2_fifo_wait(); + CHNREG32(i,0) = 0x8000; + CHNREG32(i,20) = 0x1f; + } + + SNDREG32(0x2800) = 0x000f; + g2_fifo_wait(); + G2_UNLOCK(old); +} + +/* Translates a volume from linear form to logarithmic form (required by + the AICA chip */ +/* int logs[] = { + +0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103, +105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127, +129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145, +146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159, +160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171, +172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182, +182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191, +191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199, +200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207, +208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215, +215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222, +222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228, +228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234, +234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240, +240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245, +246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251, +251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255 + +}; */ + +const static unsigned char logs[] = { + 0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61, + 63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88, + 90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106, + 108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121, + 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134, + 135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146, + 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156, + 157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167, + 167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176, + 177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185, + 186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194, + 195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202, + 203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210, + 211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218, + 219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225, + 226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233, + 233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240, + 240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246, + 247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255 +}; + +/* For the moment this is going to have to suffice, until we really + figure out what these mean. */ +#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f))) +#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)]) +//#define AICA_VOL(x) (0xff - logs[x&255]) + +static inline unsigned AICA_FREQ(unsigned freq) { + unsigned long freq_lo, freq_base = 5644800; + int freq_hi = 7; + + /* Need to convert frequency to floating point format + (freq_hi is exponent, freq_lo is mantissa) + Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */ + while (freq < freq_base && freq_hi > -8) { + freq_base >>= 1; + --freq_hi; + } + while (freq < freq_base && freq_hi > -8) { + freq_base >>= 1; + freq_hi--; + } + freq_lo = (freq<<10) / freq_base; + return (freq_hi << 11) | (freq_lo & 1023); +} + +/* Sets up a sound channel completely. This is generally good if you want + a quick and dirty way to play notes. If you want a more comprehensive + set of routines (more like PC wavetable cards) see below. + + ch is the channel to play on (0 - 63) + smpptr is the pointer to the sound data; if you're running off the + SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just + ptr. Basically, it's an offset into sound ram. + mode is one of the mode constants (16 bit, 8 bit, ADPCM) + nsamp is the number of samples to play (not number of bytes!) + freq is the sampling rate of the sound + vol is the volume, 0 to 0xff (0xff is louder) + pan is a panning constant -- 0 is left, 128 is center, 255 is right. + + This routine (and the similar ones) owe a lot to Marcus' sound example -- + I hadn't gotten quite this far into dissecting the individual regs yet. */ +void aica_play(int ch,int mode,unsigned long smpptr,int loopst,int loopend,int freq,int vol,int pan,int loopflag) { +/* int i; +*/ + int val; + int old = 0; + + /* Stop the channel (if it's already playing) */ + aica_stop(ch); + /* doesn't seem to be needed, but it's here just in case */ +/* + for (i=0; i<256; i++) { + asm("nop"); + asm("nop"); + asm("nop"); + asm("nop"); + } +*/ + G2_LOCK(old); + /* Envelope setup. The first of these is the loop point, + e.g., where the sample starts over when it loops. The second + is the loop end. This is the full length of the sample when + you are not looping, or the loop end point when you are (though + storing more than that is a waste of memory if you're not doing + volume enveloping). */ + CHNREG32(ch, 8) = loopst & 0xffff; + CHNREG32(ch, 12) = loopend & 0xffff; + + /* Write resulting values */ + CHNREG32(ch, 24) = AICA_FREQ(freq); + + /* Set volume, pan, and some other things that we don't know what + they do =) */ + CHNREG32(ch, 36) = AICA_PAN(pan) | (0xf<<8); + /* Convert the incoming volume and pan into hardware values */ + /* Vol starts at zero so we can ramp */ + vol = AICA_VOL(vol); + CHNREG32(ch, 40) = 0x24 | (vol<<8); + /* Convert the incoming volume and pan into hardware values */ + /* Vol starts at zero so we can ramp */ + + /* If we supported volume envelopes (which we don't yet) then + this value would set that up. The top 4 bits determine the + envelope speed. f is the fastest, 1 is the slowest, and 0 + seems to be an invalid value and does weird things). The + default (below) sets it into normal mode (play and terminate/loop). + CHNREG32(ch, 16) = 0xf010; + */ + CHNREG32(ch, 16) = 0x1f; /* No volume envelope */ + + + /* Set sample format, buffer address, and looping control. If + 0x0200 mask is set on reg 0, the sample loops infinitely. If + it's not set, the sample plays once and terminates. We'll + also set the bits to start playback here. */ + CHNREG32(ch, 4) = smpptr & 0xffff; + val = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16); + if (loopflag) val|=0x200; + + CHNREG32(ch, 0) = val; + + G2_UNLOCK(old); + + /* Enable playback */ + /* CHNREG32(ch, 0) |= 0xc000; */ + g2_fifo_wait(); + +#if 0 + for (i=0xff; i>=vol; i--) { + if ((i&7)==0) g2_fifo_wait(); + CHNREG32(ch, 40) = 0x24 | (i<<8);; + } + + g2_fifo_wait(); +#endif +} + +/* Stop the sound on a given channel */ +void aica_stop(int ch) { + g2_write_32(CHNREGADDR(ch, 0),(g2_read_32(CHNREGADDR(ch, 0)) & ~0x4000) | 0x8000); + g2_fifo_wait(); +} + + +/* The rest of these routines can change the channel in mid-stride so you + can do things like vibrato and panning effects. */ + +/* Set channel volume */ +void aica_vol(int ch,int vol) { +// g2_write_8(CHNREGADDR(ch, 41),AICA_VOL(vol)); + g2_write_32(CHNREGADDR(ch, 40),(g2_read_32(CHNREGADDR(ch, 40))&0xffff00ff)|(AICA_VOL(vol)<<8) ); + g2_fifo_wait(); +} + +/* Set channel pan */ +void aica_pan(int ch,int pan) { +// g2_write_8(CHNREGADDR(ch, 36),AICA_PAN(pan)); + g2_write_32(CHNREGADDR(ch, 36),(g2_read_32(CHNREGADDR(ch, 36))&0xffffff00)|(AICA_PAN(pan)) ); + g2_fifo_wait(); +} + +/* Set channel frequency */ +void aica_freq(int ch,int freq) { + g2_write_32(CHNREGADDR(ch, 24),AICA_FREQ(freq)); + g2_fifo_wait(); +} + +/* Get channel position */ +int aica_get_pos(int ch) { +#if 1 + /* Observe channel ch */ + g2_write_32(SNDREGADDR(0x280c),(g2_read_32(SNDREGADDR(0x280c))&0xffff00ff) | (ch<<8)); + g2_fifo_wait(); + /* Update position counters */ + return g2_read_32(SNDREGADDR(0x2814)) & 0xffff; +#else + /* Observe channel ch */ + g2_write_8(SNDREGADDR(0x280d),ch); + /* Update position counters */ + return g2_read_32(SNDREGADDR(0x2814)) & 0xffff; +#endif +} diff --git a/apps/plugins/sdl/src/audio/dc/aica.h b/apps/plugins/sdl/src/audio/dc/aica.h new file mode 100644 index 0000000000..2721e42821 --- /dev/null +++ b/apps/plugins/sdl/src/audio/dc/aica.h @@ -0,0 +1,40 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _AICA_H_ +#define _AICA_H_ + +#define AICA_MEM 0xa0800000 + +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +void aica_play(int ch,int mode,unsigned long smpptr,int looptst,int loopend,int freq,int vol,int pan,int loopflag); +void aica_stop(int ch); +void aica_vol(int ch,int vol); +void aica_pan(int ch,int pan); +void aica_freq(int ch,int freq); +int aica_get_pos(int ch); + +#endif diff --git a/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c b/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c new file mode 100644 index 0000000000..c45d3f8c05 --- /dev/null +++ b/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c @@ -0,0 +1,186 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org + + This file written by Ryan C. Gordon (icculus@icculus.org) +*/ +#include "SDL_config.h" + +/* Output raw audio data to a file. */ + +#if HAVE_STDIO_H +#include +#endif + +#include "SDL_rwops.h" +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_diskaudio.h" + +/* The tag name used by DISK audio */ +#define DISKAUD_DRIVER_NAME "disk" + +/* environment variables and defaults. */ +#define DISKENVR_OUTFILE "SDL_DISKAUDIOFILE" +#define DISKDEFAULT_OUTFILE "/sdlaudio.raw" +#define DISKENVR_WRITEDELAY "SDL_DISKAUDIODELAY" +#define DISKDEFAULT_WRITEDELAY 150 + +/* Audio driver functions */ +static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DISKAUD_WaitAudio(_THIS); +static void DISKAUD_PlayAudio(_THIS); +static Uint8 *DISKAUD_GetAudioBuf(_THIS); +static void DISKAUD_CloseAudio(_THIS); + +static const char *DISKAUD_GetOutputFilename(void) +{ + const char *envr = SDL_getenv(DISKENVR_OUTFILE); + return((envr != NULL) ? envr : DISKDEFAULT_OUTFILE); +} + +/* Audio driver bootstrap functions */ +static int DISKAUD_Available(void) +{ + //const char *envr = SDL_getenv("SDL_AUDIODRIVER"); +// if (envr && (SDL_strcmp(envr, DISKAUD_DRIVER_NAME) == 0)) { + return(1); +// } +// return(0); +} + +static void DISKAUD_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *DISKAUD_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + const char *envr; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + envr = SDL_getenv(DISKENVR_WRITEDELAY); + this->hidden->write_delay = (envr) ? SDL_atoi(envr) : DISKDEFAULT_WRITEDELAY; + + /* Set the function pointers */ + this->OpenAudio = DISKAUD_OpenAudio; + this->WaitAudio = DISKAUD_WaitAudio; + this->PlayAudio = DISKAUD_PlayAudio; + this->GetAudioBuf = DISKAUD_GetAudioBuf; + this->CloseAudio = DISKAUD_CloseAudio; + + this->free = DISKAUD_DeleteDevice; + + return this; +} + +AudioBootStrap DISKAUD_bootstrap = { + DISKAUD_DRIVER_NAME, "direct-to-disk audio", + DISKAUD_Available, DISKAUD_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DISKAUD_WaitAudio(_THIS) +{ + SDL_Delay(this->hidden->write_delay); +} + +static void DISKAUD_PlayAudio(_THIS) +{ + int written; + + /* Write the audio data */ + written = SDL_RWwrite(this->hidden->output, + this->hidden->mixbuf, 1, + this->hidden->mixlen); + + /* If we couldn't write, assume fatal error for now */ + if ( (Uint32)written != this->hidden->mixlen ) { + this->enabled = 0; + } +#ifdef DEBUG_AUDIO + fprintf(stderr, "Wrote %d bytes of audio data\n", written); +#endif +} + +static Uint8 *DISKAUD_GetAudioBuf(_THIS) +{ + return(this->hidden->mixbuf); +} + +static void DISKAUD_CloseAudio(_THIS) +{ + if ( this->hidden->mixbuf != NULL ) { + SDL_FreeAudioMem(this->hidden->mixbuf); + this->hidden->mixbuf = NULL; + } + if ( this->hidden->output != NULL ) { + SDL_RWclose(this->hidden->output); + this->hidden->output = NULL; + } +} + +static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + const char *fname = DISKAUD_GetOutputFilename(); + + /* Open the audio device */ + this->hidden->output = SDL_RWFromFile(fname, "wb"); + if ( this->hidden->output == NULL ) { + return(-1); + } + +#if HAVE_STDIO_H + fprintf(stderr, "WARNING: You are using the SDL disk writer" + " audio driver!\n Writing to file [%s].\n", fname); +#endif + + /* Allocate mixing buffer */ + this->hidden->mixlen = spec->size; + this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); + if ( this->hidden->mixbuf == NULL ) { + return(-1); + } + SDL_memset(this->hidden->mixbuf, spec->silence, spec->size); + + /* We're ready to rock and roll. :-) */ + return(0); +} + diff --git a/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h b/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h new file mode 100644 index 0000000000..24d7c9e34d --- /dev/null +++ b/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h @@ -0,0 +1,41 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_diskaudio_h +#define _SDL_diskaudio_h + +#include "SDL_rwops.h" +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + SDL_RWops *output; + Uint8 *mixbuf; + Uint32 mixlen; + Uint32 write_delay; +}; + +#endif /* _SDL_diskaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c b/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c new file mode 100644 index 0000000000..39f81d90ce --- /dev/null +++ b/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c @@ -0,0 +1,455 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include +#include /* For strerror() */ +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H +/* This is installed on some systems */ +#include +#else +/* This is recommended by OSS */ +#include +#endif + +#ifndef MAP_FAILED +#define MAP_FAILED ((Uint8 *)-1) +#endif + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_dmaaudio.h" + +/* The tag name used by DMA audio */ +#define DMA_DRIVER_NAME "dma" + +/* Open the audio device for playback, and don't block if busy */ +#define OPEN_FLAGS (O_RDWR|O_NONBLOCK) + +/* Audio driver functions */ +static int DMA_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DMA_WaitAudio(_THIS); +static void DMA_PlayAudio(_THIS); +static Uint8 *DMA_GetAudioBuf(_THIS); +static void DMA_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + int available; + int fd; + + available = 0; + + fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0); + if ( fd >= 0 ) { + int caps; + struct audio_buf_info info; + + if ( (ioctl(fd, SNDCTL_DSP_GETCAPS, &caps) == 0) && + (caps & DSP_CAP_TRIGGER) && (caps & DSP_CAP_MMAP) && + (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info) == 0) ) { + available = 1; + } + close(fd); + } + return(available); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + audio_fd = -1; + + /* Set the function pointers */ + this->OpenAudio = DMA_OpenAudio; + this->WaitAudio = DMA_WaitAudio; + this->PlayAudio = DMA_PlayAudio; + this->GetAudioBuf = DMA_GetAudioBuf; + this->CloseAudio = DMA_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap DMA_bootstrap = { + DMA_DRIVER_NAME, "OSS /dev/dsp DMA audio", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DMA_WaitAudio(_THIS) +{ + fd_set fdset; + + /* Check to see if the thread-parent process is still alive */ + { static int cnt = 0; + /* Note that this only works with thread implementations + that use a different process id for each thread. + */ + if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ + if ( kill(parent, 0) < 0 ) { + this->enabled = 0; + } + } + } + + /* See if we need to use timed audio synchronization */ + if ( frame_ticks ) { + /* Use timer for general audio synchronization */ + Sint32 ticks; + + ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; + if ( ticks > 0 ) { + SDL_Delay(ticks); + } + } else { + /* Use select() for audio synchronization */ + struct timeval timeout; + FD_ZERO(&fdset); + FD_SET(audio_fd, &fdset); + timeout.tv_sec = 10; + timeout.tv_usec = 0; +#ifdef DEBUG_AUDIO + fprintf(stderr, "Waiting for audio to get ready\n"); +#endif + if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) { + const char *message = +#ifdef AUDIO_OSPACE_HACK + "Audio timeout - buggy audio driver? (trying ospace)"; +#else + "Audio timeout - buggy audio driver? (disabled)"; +#endif + /* 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); +#ifdef AUDIO_OSPACE_HACK + /* We may be able to use GET_OSPACE trick */ + frame_ticks = (float)(this->spec->samples*1000) / + this->spec->freq; + next_frame = SDL_GetTicks()+frame_ticks; +#else + this->enabled = 0; + /* Don't try to close - may hang */ + audio_fd = -1; +#ifdef DEBUG_AUDIO + fprintf(stderr, "Done disabling audio\n"); +#endif +#endif /* AUDIO_OSPACE_HACK */ + } +#ifdef DEBUG_AUDIO + fprintf(stderr, "Ready!\n"); +#endif + } +} + +static void DMA_PlayAudio(_THIS) +{ + /* If timer synchronization is enabled, set the next write frame */ + if ( frame_ticks ) { + next_frame += frame_ticks; + } + return; +} + +static Uint8 *DMA_GetAudioBuf(_THIS) +{ + count_info info; + int playing; + int filling; + + /* Get number of blocks, looping if we're not using select() */ + do { + if ( ioctl(audio_fd, SNDCTL_DSP_GETOPTR, &info) < 0 ) { + /* Uh oh... */ + this->enabled = 0; + return(NULL); + } + } while ( frame_ticks && (info.blocks < 1) ); +#ifdef DEBUG_AUDIO + if ( info.blocks > 1 ) { + printf("Warning: audio underflow (%d frags)\n", info.blocks-1); + } +#endif + playing = info.ptr / this->spec.size; + filling = (playing + 1)%num_buffers; + return (dma_buf + (filling * this->spec.size)); +} + +static void DMA_CloseAudio(_THIS) +{ + if ( dma_buf != NULL ) { + munmap(dma_buf, dma_len); + dma_buf = NULL; + } + if ( audio_fd >= 0 ) { + close(audio_fd); + audio_fd = -1; + } +} + +static int DMA_ReopenAudio(_THIS, const char *audiodev, int format, int stereo, + SDL_AudioSpec *spec) +{ + int frag_spec; + int value; + + /* Close and then reopen the audio device */ + close(audio_fd); + audio_fd = open(audiodev, O_RDWR, 0); + if ( audio_fd < 0 ) { + SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); + return(-1); + } + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Determine the power of two of the fragment size */ + for ( frag_spec = 0; (0x01<size; ++frag_spec ); + if ( (0x01<size ) { + SDL_SetError("Fragment size must be a power of two"); + return(-1); + } + + /* Set the audio buffering parameters */ + if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) { + SDL_SetError("Couldn't set audio fragment spec"); + return(-1); + } + + /* Set the audio format */ + value = format; + if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) || + (value != format) ) { + SDL_SetError("Couldn't set audio format"); + return(-1); + } + + /* Set mono or stereo audio */ + value = (spec->channels > 1); + if ( (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) < 0) || + (value != stereo) ) { + SDL_SetError("Couldn't set audio channels"); + return(-1); + } + + /* Set the DSP frequency */ + value = spec->freq; + if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) { + SDL_SetError("Couldn't set audio frequency"); + return(-1); + } + spec->freq = value; + + /* We successfully re-opened the audio */ + return(0); +} + +static int DMA_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + char audiodev[1024]; + int format; + int stereo; + int value; + Uint16 test_format; + struct audio_buf_info info; + + /* Reset the timer synchronization flag */ + frame_ticks = 0.0; + + /* Open the audio device */ + audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0); + if ( audio_fd < 0 ) { + SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); + return(-1); + } + dma_buf = NULL; + ioctl(audio_fd, SNDCTL_DSP_RESET, 0); + + /* Get a list of supported hardware formats */ + if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) { + SDL_SetError("Couldn't get audio format list"); + return(-1); + } + + /* Try for a closest match on audio format */ + format = 0; + for ( test_format = SDL_FirstAudioFormat(spec->format); + ! format && test_format; ) { +#ifdef DEBUG_AUDIO + fprintf(stderr, "Trying format 0x%4.4x\n", test_format); +#endif + switch ( test_format ) { + case AUDIO_U8: + if ( value & AFMT_U8 ) { + format = AFMT_U8; + } + break; + case AUDIO_S8: + if ( value & AFMT_S8 ) { + format = AFMT_S8; + } + break; + case AUDIO_S16LSB: + if ( value & AFMT_S16_LE ) { + format = AFMT_S16_LE; + } + break; + case AUDIO_S16MSB: + if ( value & AFMT_S16_BE ) { + format = AFMT_S16_BE; + } + break; + case AUDIO_U16LSB: + if ( value & AFMT_U16_LE ) { + format = AFMT_U16_LE; + } + break; + case AUDIO_U16MSB: + if ( value & AFMT_U16_BE ) { + format = AFMT_U16_BE; + } + break; + default: + format = 0; + break; + } + if ( ! format ) { + test_format = SDL_NextAudioFormat(); + } + } + if ( format == 0 ) { + SDL_SetError("Couldn't find any hardware audio formats"); + return(-1); + } + spec->format = test_format; + + /* Set the audio format */ + value = format; + if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) || + (value != format) ) { + SDL_SetError("Couldn't set audio format"); + return(-1); + } + + /* Set mono or stereo audio (currently only two channels supported) */ + stereo = (spec->channels > 1); + ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo); + if ( stereo ) { + spec->channels = 2; + } else { + spec->channels = 1; + } + + /* Because some drivers don't allow setting the buffer size + after setting the format, we must re-open the audio device + once we know what format and channels are supported + */ + if ( DMA_ReopenAudio(this, audiodev, format, stereo, spec) < 0 ) { + /* Error is set by DMA_ReopenAudio() */ + return(-1); + } + + /* Memory map the audio buffer */ + if ( ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info) < 0 ) { + SDL_SetError("Couldn't get OSPACE parameters"); + return(-1); + } + spec->size = info.fragsize; + spec->samples = spec->size / ((spec->format & 0xFF) / 8); + spec->samples /= spec->channels; + num_buffers = info.fragstotal; + dma_len = num_buffers*spec->size; + dma_buf = (Uint8 *)mmap(NULL, dma_len, PROT_WRITE, MAP_SHARED, + audio_fd, 0); + if ( dma_buf == MAP_FAILED ) { + SDL_SetError("DMA memory map failed"); + dma_buf = NULL; + return(-1); + } + SDL_memset(dma_buf, spec->silence, dma_len); + + /* Check to see if we need to use select() workaround */ + { char *workaround; + workaround = SDL_getenv("SDL_DSP_NOSELECT"); + if ( workaround ) { + frame_ticks = (float)(spec->samples*1000)/spec->freq; + next_frame = SDL_GetTicks()+frame_ticks; + } + } + + /* Trigger audio playback */ + value = 0; + ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value); + value = PCM_ENABLE_OUTPUT; + if ( ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value) < 0 ) { + SDL_SetError("Couldn't trigger audio output"); + return(-1); + } + + /* Get the parent process id (we're the parent of the audio thread) */ + parent = getpid(); + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h b/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h new file mode 100644 index 0000000000..9a45f732a1 --- /dev/null +++ b/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h @@ -0,0 +1,59 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_dspaudio_h +#define _SDL_dspaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + int audio_fd; + + /* The parent process id, to detect when application quits */ + pid_t parent; + + /* Raw mixing buffer */ + Uint8 *dma_buf; + int dma_len; + int num_buffers; + + /* Support for audio timing using a timer, in addition to select() */ + float frame_ticks; + float next_frame; +}; +#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */ + +/* Old variable names */ +#define audio_fd (this->hidden->audio_fd) +#define parent (this->hidden->parent) +#define dma_buf (this->hidden->dma_buf) +#define dma_len (this->hidden->dma_len) +#define num_buffers (this->hidden->num_buffers) +#define frame_ticks (this->hidden->frame_ticks) +#define next_frame (this->hidden->next_frame) + +#endif /* _SDL_dspaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c b/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c new file mode 100644 index 0000000000..1dcd2421ec --- /dev/null +++ b/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c @@ -0,0 +1,242 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer (For IRIX 6.5 and higher) */ +/* patch for IRIX 5 by Georg Schwarz 18/07/2004 */ + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "SDL_irixaudio.h" + + +#ifndef AL_RESOURCE /* as a test whether we use the old IRIX audio libraries */ +#define OLD_IRIX_AUDIO +#define alClosePort(x) ALcloseport(x) +#define alFreeConfig(x) ALfreeconfig(x) +#define alGetFillable(x) ALgetfillable(x) +#define alNewConfig() ALnewconfig() +#define alOpenPort(x,y,z) ALopenport(x,y,z) +#define alSetChannels(x,y) ALsetchannels(x,y) +#define alSetQueueSize(x,y) ALsetqueuesize(x,y) +#define alSetSampFmt(x,y) ALsetsampfmt(x,y) +#define alSetWidth(x,y) ALsetwidth(x,y) +#endif + +/* Audio driver functions */ +static int AL_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void AL_WaitAudio(_THIS); +static void AL_PlayAudio(_THIS); +static Uint8 *AL_GetAudioBuf(_THIS); +static void AL_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + return 1; +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = AL_OpenAudio; + this->WaitAudio = AL_WaitAudio; + this->PlayAudio = AL_PlayAudio; + this->GetAudioBuf = AL_GetAudioBuf; + this->CloseAudio = AL_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap DMEDIA_bootstrap = { + "AL", "IRIX DMedia audio", + Audio_Available, Audio_CreateDevice +}; + + +void static AL_WaitAudio(_THIS) +{ + Sint32 timeleft; + + timeleft = this->spec.samples - alGetFillable(audio_port); + if ( timeleft > 0 ) { + timeleft /= (this->spec.freq/1000); + SDL_Delay((Uint32)timeleft); + } +} + +static void AL_PlayAudio(_THIS) +{ + /* Write the audio data out */ + if ( alWriteFrames(audio_port, mixbuf, this->spec.samples) < 0 ) { + /* Assume fatal error, for now */ + this->enabled = 0; + } +} + +static Uint8 *AL_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void AL_CloseAudio(_THIS) +{ + if ( mixbuf != NULL ) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if ( audio_port != NULL ) { + alClosePort(audio_port); + audio_port = NULL; + } +} + +static int AL_OpenAudio(_THIS, SDL_AudioSpec * spec) +{ + Uint16 test_format = SDL_FirstAudioFormat(spec->format); + long width = 0; + long fmt = 0; + int valid = 0; + +#ifdef OLD_IRIX_AUDIO + { + long audio_param[2]; + audio_param[0] = AL_OUTPUT_RATE; + audio_param[1] = spec->freq; + valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0); + } +#else + { + ALpv audio_param; + audio_param.param = AL_RATE; + audio_param.value.i = spec->freq; + valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0); + } +#endif + + while ((!valid) && (test_format)) { + valid = 1; + spec->format = test_format; + + switch (test_format) { + case AUDIO_S8: + width = AL_SAMPLE_8; + fmt = AL_SAMPFMT_TWOSCOMP; + break; + + case AUDIO_S16SYS: + width = AL_SAMPLE_16; + fmt = AL_SAMPFMT_TWOSCOMP; + break; + + default: + valid = 0; + test_format = SDL_NextAudioFormat(); + break; + } + + if (valid) { + ALconfig audio_config = alNewConfig(); + valid = 0; + if (audio_config) { + if (alSetChannels(audio_config, spec->channels) < 0) { + if (spec->channels > 2) { /* can't handle > stereo? */ + spec->channels = 2; /* try again below. */ + } + } + + if ((alSetSampFmt(audio_config, fmt) >= 0) && + ((!width) || (alSetWidth(audio_config, width) >= 0)) && + (alSetQueueSize(audio_config, spec->samples * 2) >= 0) && + (alSetChannels(audio_config, spec->channels) >= 0)) { + + audio_port = alOpenPort("SDL audio", "w", audio_config); + if (audio_port == NULL) { + /* docs say AL_BAD_CHANNELS happens here, too. */ + int err = oserror(); + if (err == AL_BAD_CHANNELS) { + spec->channels = 2; + alSetChannels(audio_config, spec->channels); + audio_port = alOpenPort("SDL audio", "w", + audio_config); + } + } + + if (audio_port != NULL) { + valid = 1; + } + } + + alFreeConfig(audio_config); + } + } + } + + if (!valid) { + SDL_SetError("Unsupported audio format"); + return (-1); + } + + /* Update the fragment size as size in bytes */ + SDL_CalculateAudioSpec(spec); + + /* Allocate mixing buffer */ + mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size); + if (mixbuf == NULL) { + SDL_OutOfMemory(); + return (-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* We're ready to rock and roll. :-) */ + return (0); +} + diff --git a/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h b/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h new file mode 100644 index 0000000000..c04f497cea --- /dev/null +++ b/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h @@ -0,0 +1,45 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_lowaudio_h +#define _SDL_lowaudio_h + +#include + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the audio functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The handle for the audio device */ + ALport audio_port; + + Uint8 *mixbuf; /* The app mixing buffer */ +}; + +/* Old variable names */ +#define audio_port (this->hidden->audio_port) +#define mixbuf (this->hidden->mixbuf) + +#endif /* _SDL_lowaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c new file mode 100644 index 0000000000..256c547f9b --- /dev/null +++ b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c @@ -0,0 +1,340 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org + + Modified in Oct 2004 by Hannu Savolainen + hannu@opensound.com +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include /* For perror() */ +#include /* For strerror() */ +#include +#include +#include +#include +#include +#include +#include + +#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H +/* This is installed on some systems */ +#include +#else +/* This is recommended by OSS */ +#include +#endif + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_dspaudio.h" + +/* The tag name used by DSP audio */ +#define DSP_DRIVER_NAME "dsp" + +/* Open the audio device for playback, and don't block if busy */ +#define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) + +/* Audio driver functions */ +static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DSP_WaitAudio(_THIS); +static void DSP_PlayAudio(_THIS); +static Uint8 *DSP_GetAudioBuf(_THIS); +static void DSP_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + int fd; + int available; + + available = 0; + fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0); + if ( fd >= 0 ) { + available = 1; + close(fd); + } + return(available); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + audio_fd = -1; + + /* Set the function pointers */ + this->OpenAudio = DSP_OpenAudio; + this->WaitAudio = DSP_WaitAudio; + this->PlayAudio = DSP_PlayAudio; + this->GetAudioBuf = DSP_GetAudioBuf; + this->CloseAudio = DSP_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap DSP_bootstrap = { + DSP_DRIVER_NAME, "OSS /dev/dsp standard audio", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DSP_WaitAudio(_THIS) +{ + /* Not needed at all since OSS handles waiting automagically */ +} + +static void DSP_PlayAudio(_THIS) +{ + if (write(audio_fd, mixbuf, mixlen)==-1) + { + perror("Audio write"); + this->enabled = 0; + } + +#ifdef DEBUG_AUDIO + fprintf(stderr, "Wrote %d bytes of audio data\n", mixlen); +#endif +} + +static Uint8 *DSP_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void DSP_CloseAudio(_THIS) +{ + if ( mixbuf != NULL ) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if ( audio_fd >= 0 ) { + close(audio_fd); + audio_fd = -1; + } +} + +static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + char audiodev[1024]; + int format; + int value; + int frag_spec; + Uint16 test_format; + + /* Make sure fragment size stays a power of 2, or OSS fails. */ + /* I don't know which of these are actually legal values, though... */ + if (spec->channels > 8) + spec->channels = 8; + else if (spec->channels > 4) + spec->channels = 4; + else if (spec->channels > 2) + spec->channels = 2; + + /* Open the audio device */ + audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0); + if ( audio_fd < 0 ) { + SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); + return(-1); + } + mixbuf = NULL; + + /* Make the file descriptor use blocking writes with fcntl() */ + { long flags; + flags = fcntl(audio_fd, F_GETFL); + flags &= ~O_NONBLOCK; + if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) { + SDL_SetError("Couldn't set audio blocking mode"); + DSP_CloseAudio(this); + return(-1); + } + } + + /* Get a list of supported hardware formats */ + if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) { + perror("SNDCTL_DSP_GETFMTS"); + SDL_SetError("Couldn't get audio format list"); + DSP_CloseAudio(this); + return(-1); + } + + /* Try for a closest match on audio format */ + format = 0; + for ( test_format = SDL_FirstAudioFormat(spec->format); + ! format && test_format; ) { +#ifdef DEBUG_AUDIO + fprintf(stderr, "Trying format 0x%4.4x\n", test_format); +#endif + switch ( test_format ) { + case AUDIO_U8: + if ( value & AFMT_U8 ) { + format = AFMT_U8; + } + break; + case AUDIO_S16LSB: + if ( value & AFMT_S16_LE ) { + format = AFMT_S16_LE; + } + break; + case AUDIO_S16MSB: + if ( value & AFMT_S16_BE ) { + format = AFMT_S16_BE; + } + break; +#if 0 +/* + * These formats are not used by any real life systems so they are not + * needed here. + */ + case AUDIO_S8: + if ( value & AFMT_S8 ) { + format = AFMT_S8; + } + break; + case AUDIO_U16LSB: + if ( value & AFMT_U16_LE ) { + format = AFMT_U16_LE; + } + break; + case AUDIO_U16MSB: + if ( value & AFMT_U16_BE ) { + format = AFMT_U16_BE; + } + break; +#endif + default: + format = 0; + break; + } + if ( ! format ) { + test_format = SDL_NextAudioFormat(); + } + } + if ( format == 0 ) { + SDL_SetError("Couldn't find any hardware audio formats"); + DSP_CloseAudio(this); + return(-1); + } + spec->format = test_format; + + /* Set the audio format */ + value = format; + if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) || + (value != format) ) { + perror("SNDCTL_DSP_SETFMT"); + SDL_SetError("Couldn't set audio format"); + DSP_CloseAudio(this); + return(-1); + } + + /* Set the number of channels of output */ + value = spec->channels; + if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) { + perror("SNDCTL_DSP_CHANNELS"); + SDL_SetError("Cannot set the number of channels"); + DSP_CloseAudio(this); + return(-1); + } + spec->channels = value; + + /* Set the DSP frequency */ + value = spec->freq; + if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) { + perror("SNDCTL_DSP_SPEED"); + SDL_SetError("Couldn't set audio frequency"); + DSP_CloseAudio(this); + return(-1); + } + spec->freq = value; + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Determine the power of two of the fragment size */ + for ( frag_spec = 0; (0x01U<size; ++frag_spec ); + if ( (0x01U<size ) { + SDL_SetError("Fragment size must be a power of two"); + DSP_CloseAudio(this); + return(-1); + } + frag_spec |= 0x00020000; /* two fragments, for low latency */ + + /* Set the audio buffering parameters */ +#ifdef DEBUG_AUDIO + fprintf(stderr, "Requesting %d fragments of size %d\n", + (frag_spec >> 16), 1<<(frag_spec&0xFFFF)); +#endif + if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) { + perror("SNDCTL_DSP_SETFRAGMENT"); + } +#ifdef DEBUG_AUDIO + { audio_buf_info info; + ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info); + fprintf(stderr, "fragments = %d\n", info.fragments); + fprintf(stderr, "fragstotal = %d\n", info.fragstotal); + fprintf(stderr, "fragsize = %d\n", info.fragsize); + fprintf(stderr, "bytes = %d\n", info.bytes); + } +#endif + + /* Allocate mixing buffer */ + mixlen = spec->size; + mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); + if ( mixbuf == NULL ) { + DSP_CloseAudio(this); + return(-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* Get the parent process id (we're the parent of the audio thread) */ + parent = getpid(); + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h new file mode 100644 index 0000000000..382544f967 --- /dev/null +++ b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h @@ -0,0 +1,53 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_dspaudio_h +#define _SDL_dspaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + int audio_fd; + + /* The parent process id, to detect when application quits */ + pid_t parent; + + /* Raw mixing buffer */ + Uint8 *mixbuf; + int mixlen; +}; +#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */ + +/* Old variable names */ +#define audio_fd (this->hidden->audio_fd) +#define parent (this->hidden->parent) +#define mixbuf (this->hidden->mixbuf) +#define mixlen (this->hidden->mixlen) +#define frame_ticks (this->hidden->frame_ticks) +#define next_frame (this->hidden->next_frame) + +#endif /* _SDL_dspaudio_h */ diff --git a/apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.c b/apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.c new file mode 100644 index 0000000000..484b50d40c --- /dev/null +++ b/apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.c @@ -0,0 +1,156 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org + + This file written by Ryan C. Gordon (icculus@icculus.org) +*/ +#include "SDL_config.h" + +/* Output audio to nowhere... */ + +#include "SDL_rwops.h" +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_dummyaudio.h" + +/* The tag name used by DUMMY audio */ +#define DUMMYAUD_DRIVER_NAME "dummy" + +/* Audio driver functions */ +static int DUMMYAUD_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DUMMYAUD_WaitAudio(_THIS); +static void DUMMYAUD_PlayAudio(_THIS); +static Uint8 *DUMMYAUD_GetAudioBuf(_THIS); +static void DUMMYAUD_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ +static int DUMMYAUD_Available(void) +{ + const char *envr = SDL_getenv("SDL_AUDIODRIVER"); + if (envr && (SDL_strcmp(envr, DUMMYAUD_DRIVER_NAME) == 0)) { + return(1); + } + return(0); +} + +static void DUMMYAUD_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *DUMMYAUD_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = DUMMYAUD_OpenAudio; + this->WaitAudio = DUMMYAUD_WaitAudio; + this->PlayAudio = DUMMYAUD_PlayAudio; + this->GetAudioBuf = DUMMYAUD_GetAudioBuf; + this->CloseAudio = DUMMYAUD_CloseAudio; + + this->free = DUMMYAUD_DeleteDevice; + + return this; +} + +AudioBootStrap DUMMYAUD_bootstrap = { + DUMMYAUD_DRIVER_NAME, "SDL dummy audio driver", + DUMMYAUD_Available, DUMMYAUD_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DUMMYAUD_WaitAudio(_THIS) +{ + /* Don't block on first calls to simulate initial fragment filling. */ + if (this->hidden->initial_calls) + this->hidden->initial_calls--; + else + SDL_Delay(this->hidden->write_delay); +} + +static void DUMMYAUD_PlayAudio(_THIS) +{ + /* no-op...this is a null driver. */ +} + +static Uint8 *DUMMYAUD_GetAudioBuf(_THIS) +{ + return(this->hidden->mixbuf); +} + +static void DUMMYAUD_CloseAudio(_THIS) +{ + if ( this->hidden->mixbuf != NULL ) { + SDL_FreeAudioMem(this->hidden->mixbuf); + this->hidden->mixbuf = NULL; + } +} + +static int DUMMYAUD_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + float bytes_per_sec = 0.0f; + + /* Allocate mixing buffer */ + this->hidden->mixlen = spec->size; + this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); + if ( this->hidden->mixbuf == NULL ) { + return(-1); + } + SDL_memset(this->hidden->mixbuf, spec->silence, spec->size); + + bytes_per_sec = (float) (((spec->format & 0xFF) / 8) * + spec->channels * spec->freq); + + /* + * We try to make this request more audio at the correct rate for + * a given audio spec, so timing stays fairly faithful. + * Also, we have it not block at all for the first two calls, so + * it seems like we're filling two audio fragments right out of the + * gate, like other SDL drivers tend to do. + */ + this->hidden->initial_calls = 2; + this->hidden->write_delay = + (Uint32) ((((float) spec->size) / bytes_per_sec) * 1000.0f); + + /* We're ready to rock and roll. :-) */ + return(0); +} + diff --git a/apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.h b/apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.h new file mode 100644 index 0000000000..74a69cab7b --- /dev/null +++ b/apps/plugins/sdl/src/audio/dummy/SDL_dummyaudio.h @@ -0,0 +1,40 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_dummyaudio_h +#define _SDL_dummyaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + Uint8 *mixbuf; + Uint32 mixlen; + Uint32 write_delay; + Uint32 initial_calls; +}; + +#endif /* _SDL_dummyaudio_h */ diff --git a/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c b/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c new file mode 100644 index 0000000000..f54b0ea9c5 --- /dev/null +++ b/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c @@ -0,0 +1,323 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to an ESD network stream mixing buffer */ + +#include +#include +#include +#include +#include + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_esdaudio.h" + +#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC +#include "SDL_name.h" +#include "SDL_loadso.h" +#else +#define SDL_NAME(X) X +#endif + +/* The tag name used by ESD audio */ +#define ESD_DRIVER_NAME "esd" + +/* Audio driver functions */ +static int ESD_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void ESD_WaitAudio(_THIS); +static void ESD_PlayAudio(_THIS); +static Uint8 *ESD_GetAudioBuf(_THIS); +static void ESD_CloseAudio(_THIS); + +#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC + +static const char *esd_library = SDL_AUDIO_DRIVER_ESD_DYNAMIC; +static void *esd_handle = NULL; +static int esd_loaded = 0; + +static int (*SDL_NAME(esd_open_sound))( const char *host ); +static int (*SDL_NAME(esd_close))( int esd ); +static int (*SDL_NAME(esd_play_stream))( esd_format_t format, int rate, + const char *host, const char *name ); +static struct { + const char *name; + void **func; +} esd_functions[] = { + { "esd_open_sound", (void **)&SDL_NAME(esd_open_sound) }, + { "esd_close", (void **)&SDL_NAME(esd_close) }, + { "esd_play_stream", (void **)&SDL_NAME(esd_play_stream) }, +}; + +static void UnloadESDLibrary() +{ + if ( esd_loaded ) { + SDL_UnloadObject(esd_handle); + esd_handle = NULL; + esd_loaded = 0; + } +} + +static int LoadESDLibrary(void) +{ + int i, retval = -1; + + esd_handle = SDL_LoadObject(esd_library); + if ( esd_handle ) { + esd_loaded = 1; + retval = 0; + for ( i=0; i= 0 ) { + available = 1; + SDL_NAME(esd_close)(connection); + } + UnloadESDLibrary(); + return(available); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); + UnloadESDLibrary(); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + LoadESDLibrary(); + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + audio_fd = -1; + + /* Set the function pointers */ + this->OpenAudio = ESD_OpenAudio; + this->WaitAudio = ESD_WaitAudio; + this->PlayAudio = ESD_PlayAudio; + this->GetAudioBuf = ESD_GetAudioBuf; + this->CloseAudio = ESD_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap ESD_bootstrap = { + ESD_DRIVER_NAME, "Enlightened Sound Daemon", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void ESD_WaitAudio(_THIS) +{ + Sint32 ticks; + + /* Check to see if the thread-parent process is still alive */ + { static int cnt = 0; + /* Note that this only works with thread implementations + that use a different process id for each thread. + */ + if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ + if ( kill(parent, 0) < 0 ) { + this->enabled = 0; + } + } + } + + /* Use timer for general audio synchronization */ + ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; + if ( ticks > 0 ) { + SDL_Delay(ticks); + } +} + +static void ESD_PlayAudio(_THIS) +{ + int written; + + /* Write the audio data, checking for EAGAIN on broken audio drivers */ + do { + written = write(audio_fd, mixbuf, mixlen); + if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) { + SDL_Delay(1); /* Let a little CPU time go by */ + } + } while ( (written < 0) && + ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) ); + + /* Set the next write frame */ + next_frame += frame_ticks; + + /* If we couldn't write, assume fatal error for now */ + if ( written < 0 ) { + this->enabled = 0; + } +} + +static Uint8 *ESD_GetAudioBuf(_THIS) +{ + return(mixbuf); +} + +static void ESD_CloseAudio(_THIS) +{ + if ( mixbuf != NULL ) { + SDL_FreeAudioMem(mixbuf); + mixbuf = NULL; + } + if ( audio_fd >= 0 ) { + SDL_NAME(esd_close)(audio_fd); + audio_fd = -1; + } +} + +/* Try to get the name of the program */ +static char *get_progname(void) +{ + char *progname = NULL; +#ifdef __LINUX__ + FILE *fp; + static char temp[BUFSIZ]; + + SDL_snprintf(temp, SDL_arraysize(temp), "/proc/%d/cmdline", getpid()); + fp = fopen(temp, "r"); + if ( fp != NULL ) { + if ( fgets(temp, sizeof(temp)-1, fp) ) { + progname = SDL_strrchr(temp, '/'); + if ( progname == NULL ) { + progname = temp; + } else { + progname = progname+1; + } + } + fclose(fp); + } +#endif + return(progname); +} + +static int ESD_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + esd_format_t format; + + /* Convert audio spec to the ESD audio format */ + format = (ESD_STREAM | ESD_PLAY); + switch ( spec->format & 0xFF ) { + case 8: + format |= ESD_BITS8; + break; + case 16: + format |= ESD_BITS16; + break; + default: + SDL_SetError("Unsupported ESD audio format"); + return(-1); + } + if ( spec->channels == 1 ) { + format |= ESD_MONO; + } else { + format |= ESD_STEREO; + } +#if 0 + spec->samples = ESD_BUF_SIZE; /* Darn, no way to change this yet */ +#endif + + /* Open a connection to the ESD audio server */ + audio_fd = SDL_NAME(esd_play_stream)(format, spec->freq, NULL, get_progname()); + if ( audio_fd < 0 ) { + SDL_SetError("Couldn't open ESD connection"); + return(-1); + } + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + frame_ticks = (float)(spec->samples*1000)/spec->freq; + next_frame = SDL_GetTicks()+frame_ticks; + + /* Allocate mixing buffer */ + mixlen = spec->size; + mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); + if ( mixbuf == NULL ) { + return(-1); + } + SDL_memset(mixbuf, spec->silence, spec->size); + + /* Get the parent process id (we're the parent of the audio thread) */ + parent = getpid(); + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h b/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h new file mode 100644 index 0000000000..da4ae6a04b --- /dev/null +++ b/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h @@ -0,0 +1,57 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_esdaudio_h +#define _SDL_esdaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + int audio_fd; + + /* The parent process id, to detect when application quits */ + pid_t parent; + + /* Raw mixing buffer */ + Uint8 *mixbuf; + int mixlen; + + /* Support for audio timing using a timer */ + float frame_ticks; + float next_frame; +}; +#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */ + +/* Old variable names */ +#define audio_fd (this->hidden->audio_fd) +#define parent (this->hidden->parent) +#define mixbuf (this->hidden->mixbuf) +#define mixlen (this->hidden->mixlen) +#define frame_ticks (this->hidden->frame_ticks) +#define next_frame (this->hidden->next_frame) + +#endif /* _SDL_esdaudio_h */ diff --git a/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c b/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c new file mode 100644 index 0000000000..31316d1fd9 --- /dev/null +++ b/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c @@ -0,0 +1,291 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#include +#include +#include +#if MAC_OS_X_VERSION_MAX_ALLOWED <= 1050 +#include +#endif + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" +#include "SDL_coreaudio.h" + + +/* Audio driver functions */ + +static int Core_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Core_WaitAudio(_THIS); +static void Core_PlayAudio(_THIS); +static Uint8 *Core_GetAudioBuf(_THIS); +static void Core_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Core_OpenAudio; + this->WaitAudio = Core_WaitAudio; + this->PlayAudio = Core_PlayAudio; + this->GetAudioBuf = Core_GetAudioBuf; + this->CloseAudio = Core_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap COREAUDIO_bootstrap = { + "coreaudio", "Mac OS X CoreAudio", + Audio_Available, Audio_CreateDevice +}; + +/* The CoreAudio callback */ +static OSStatus audioCallback (void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData) +{ + SDL_AudioDevice *this = (SDL_AudioDevice *)inRefCon; + UInt32 remaining, len; + AudioBuffer *abuf; + void *ptr; + UInt32 i; + + /* Only do anything if audio is enabled and not paused */ + if ( ! this->enabled || this->paused ) { + for (i = 0; i < ioData->mNumberBuffers; i++) { + abuf = &ioData->mBuffers[i]; + SDL_memset(abuf->mData, this->spec.silence, abuf->mDataByteSize); + } + return 0; + } + + /* No SDL conversion should be needed here, ever, since we accept + any input format in OpenAudio, and leave the conversion to CoreAudio. + */ + /* + assert(!this->convert.needed); + assert(this->spec.channels == ioData->mNumberChannels); + */ + + for (i = 0; i < ioData->mNumberBuffers; i++) { + abuf = &ioData->mBuffers[i]; + remaining = abuf->mDataByteSize; + ptr = abuf->mData; + while (remaining > 0) { + if (bufferOffset >= bufferSize) { + /* Generate the data */ + SDL_memset(buffer, this->spec.silence, bufferSize); + SDL_mutexP(this->mixer_lock); + (*this->spec.callback)(this->spec.userdata, + buffer, bufferSize); + SDL_mutexV(this->mixer_lock); + bufferOffset = 0; + } + + len = bufferSize - bufferOffset; + if (len > remaining) + len = remaining; + SDL_memcpy(ptr, (char *)buffer + bufferOffset, len); + ptr = (char *)ptr + len; + remaining -= len; + bufferOffset += len; + } + } + + return 0; +} + +/* Dummy functions -- we don't use thread-based audio */ +void Core_WaitAudio(_THIS) +{ + return; +} + +void Core_PlayAudio(_THIS) +{ + return; +} + +Uint8 *Core_GetAudioBuf(_THIS) +{ + return(NULL); +} + +void Core_CloseAudio(_THIS) +{ + OSStatus result; + struct AURenderCallbackStruct callback; + + /* stop processing the audio unit */ + result = AudioOutputUnitStop (outputAudioUnit); + if (result != noErr) { + SDL_SetError("Core_CloseAudio: AudioOutputUnitStop"); + return; + } + + /* Remove the input callback */ + callback.inputProc = 0; + callback.inputProcRefCon = 0; + result = AudioUnitSetProperty (outputAudioUnit, + kAudioUnitProperty_SetRenderCallback, + kAudioUnitScope_Input, + 0, + &callback, + sizeof(callback)); + if (result != noErr) { + SDL_SetError("Core_CloseAudio: AudioUnitSetProperty (kAudioUnitProperty_SetInputCallback)"); + return; + } + + result = CloseComponent(outputAudioUnit); + if (result != noErr) { + SDL_SetError("Core_CloseAudio: CloseComponent"); + return; + } + + SDL_free(buffer); +} + +#define CHECK_RESULT(msg) \ + if (result != noErr) { \ + SDL_SetError("Failed to start CoreAudio: " msg); \ + return -1; \ + } + + +int Core_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + OSStatus result = noErr; + Component comp; + ComponentDescription desc; + struct AURenderCallbackStruct callback; + AudioStreamBasicDescription requestedDesc; + + /* Setup a AudioStreamBasicDescription with the requested format */ + requestedDesc.mFormatID = kAudioFormatLinearPCM; + requestedDesc.mFormatFlags = kLinearPCMFormatFlagIsPacked; + requestedDesc.mChannelsPerFrame = spec->channels; + requestedDesc.mSampleRate = spec->freq; + + requestedDesc.mBitsPerChannel = spec->format & 0xFF; + if (spec->format & 0x8000) + requestedDesc.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger; + if (spec->format & 0x1000) + requestedDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + + requestedDesc.mFramesPerPacket = 1; + requestedDesc.mBytesPerFrame = requestedDesc.mBitsPerChannel * requestedDesc.mChannelsPerFrame / 8; + requestedDesc.mBytesPerPacket = requestedDesc.mBytesPerFrame * requestedDesc.mFramesPerPacket; + + + /* Locate the default output audio unit */ + desc.componentType = kAudioUnitType_Output; + desc.componentSubType = kAudioUnitSubType_DefaultOutput; + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + + comp = FindNextComponent (NULL, &desc); + if (comp == NULL) { + SDL_SetError ("Failed to start CoreAudio: FindNextComponent returned NULL"); + return -1; + } + + /* Open & initialize the default output audio unit */ + result = OpenAComponent (comp, &outputAudioUnit); + CHECK_RESULT("OpenAComponent") + + result = AudioUnitInitialize (outputAudioUnit); + CHECK_RESULT("AudioUnitInitialize") + + /* Set the input format of the audio unit. */ + result = AudioUnitSetProperty (outputAudioUnit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + 0, + &requestedDesc, + sizeof (requestedDesc)); + CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_StreamFormat)") + + /* Set the audio callback */ + callback.inputProc = audioCallback; + callback.inputProcRefCon = this; + result = AudioUnitSetProperty (outputAudioUnit, + kAudioUnitProperty_SetRenderCallback, + kAudioUnitScope_Input, + 0, + &callback, + sizeof(callback)); + CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_SetInputCallback)") + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + /* Allocate a sample buffer */ + bufferOffset = bufferSize = this->spec.size; + buffer = SDL_malloc(bufferSize); + + /* Finally, start processing of the audio unit */ + result = AudioOutputUnitStart (outputAudioUnit); + CHECK_RESULT("AudioOutputUnitStart") + + + /* We're running! */ + return(1); +} diff --git a/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h b/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h new file mode 100644 index 0000000000..c11bc03a2b --- /dev/null +++ b/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h @@ -0,0 +1,45 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_coreaudio_h +#define _SDL_coreaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + AudioUnit outputAudioUnit; + void *buffer; + UInt32 bufferOffset; + UInt32 bufferSize; +}; + +/* Old variable names */ +#define outputAudioUnit (this->hidden->outputAudioUnit) +#define buffer (this->hidden->buffer) +#define bufferOffset (this->hidden->bufferOffset) +#define bufferSize (this->hidden->bufferSize) + +#endif /* _SDL_coreaudio_h */ diff --git a/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c b/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c new file mode 100644 index 0000000000..1b3d49e198 --- /dev/null +++ b/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c @@ -0,0 +1,496 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#if defined(__APPLE__) && defined(__MACH__) +# include +#elif TARGET_API_MAC_CARBON && (UNIVERSAL_INTERFACES_VERSION > 0x0335) +# include +#else +# include /* SoundManager interface */ +# include +# include +#endif + +#if !defined(NewSndCallBackUPP) && (UNIVERSAL_INTERFACES_VERSION < 0x0335) +#if !defined(NewSndCallBackProc) /* avoid circular redefinition... */ +#define NewSndCallBackUPP NewSndCallBackProc +#endif +#if !defined(NewSndCallBackUPP) +#define NewSndCallBackUPP NewSndCallBackProc +#endif +#endif + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" +#include "SDL_romaudio.h" + +/* Audio driver functions */ + +static void Mac_CloseAudio(_THIS); +static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Mac_LockAudio(_THIS); +static void Mac_UnlockAudio(_THIS); + +/* Audio driver bootstrap functions */ + + +static int Audio_Available(void) +{ + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Mac_OpenAudio; + this->CloseAudio = Mac_CloseAudio; + this->LockAudio = Mac_LockAudio; + this->UnlockAudio = Mac_UnlockAudio; + this->free = Audio_DeleteDevice; + +#ifdef __MACOSX__ /* Mac OS X uses threaded audio, so normal thread code is okay */ + this->LockAudio = NULL; + this->UnlockAudio = NULL; +#endif + return this; +} + +AudioBootStrap SNDMGR_bootstrap = { + "sndmgr", "MacOS SoundManager 3.0", + Audio_Available, Audio_CreateDevice +}; + +#if defined(TARGET_API_MAC_CARBON) || defined(USE_RYANS_SOUNDCODE) +/* This works correctly on Mac OS X */ + +#pragma options align=power + +static volatile SInt32 audio_is_locked = 0; +static volatile SInt32 need_to_mix = 0; + +static UInt8 *buffer[2]; +static volatile UInt32 running = 0; +static CmpSoundHeader header; +static volatile Uint32 fill_me = 0; + +static void mix_buffer(SDL_AudioDevice *audio, UInt8 *buffer) +{ + if ( ! audio->paused ) { +#ifdef __MACOSX__ + SDL_mutexP(audio->mixer_lock); +#endif + if ( audio->convert.needed ) { + audio->spec.callback(audio->spec.userdata, + (Uint8 *)audio->convert.buf,audio->convert.len); + SDL_ConvertAudio(&audio->convert); + if ( audio->convert.len_cvt != audio->spec.size ) { + /* Uh oh... probably crashes here */; + } + SDL_memcpy(buffer, audio->convert.buf, audio->convert.len_cvt); + } else { + audio->spec.callback(audio->spec.userdata, buffer, audio->spec.size); + } +#ifdef __MACOSX__ + SDL_mutexV(audio->mixer_lock); +#endif + } + + DecrementAtomic((SInt32 *) &need_to_mix); +} + +static void Mac_LockAudio(_THIS) +{ + IncrementAtomic((SInt32 *) &audio_is_locked); +} + +static void Mac_UnlockAudio(_THIS) +{ + SInt32 oldval; + + oldval = DecrementAtomic((SInt32 *) &audio_is_locked); + if ( oldval != 1 ) /* != 1 means audio is still locked. */ + return; + + /* Did we miss the chance to mix in an interrupt? Do it now. */ + if ( BitAndAtomic (0xFFFFFFFF, (UInt32 *) &need_to_mix) ) { + /* + * Note that this could be a problem if you missed an interrupt + * while the audio was locked, and get preempted by a second + * interrupt here, but that means you locked for way too long anyhow. + */ + mix_buffer (this, buffer[fill_me]); + } +} + +static void callBackProc (SndChannel *chan, SndCommand *cmd_passed ) { + UInt32 play_me; + SndCommand cmd; + SDL_AudioDevice *audio = (SDL_AudioDevice *)chan->userInfo; + + IncrementAtomic((SInt32 *) &need_to_mix); + + fill_me = cmd_passed->param2; /* buffer that has just finished playing, so fill it */ + play_me = ! fill_me; /* filled buffer to play _now_ */ + + if ( ! audio->enabled ) { + return; + } + + /* queue previously mixed buffer for playback. */ + header.samplePtr = (Ptr)buffer[play_me]; + cmd.cmd = bufferCmd; + cmd.param1 = 0; + cmd.param2 = (long)&header; + SndDoCommand (chan, &cmd, 0); + + memset (buffer[fill_me], 0, audio->spec.size); + + /* + * if audio device isn't locked, mix the next buffer to be queued in + * the memory block that just finished playing. + */ + if ( ! BitAndAtomic(0xFFFFFFFF, (UInt32 *) &audio_is_locked) ) { + mix_buffer (audio, buffer[fill_me]); + } + + /* set this callback to run again when current buffer drains. */ + if ( running ) { + cmd.cmd = callBackCmd; + cmd.param1 = 0; + cmd.param2 = play_me; + + SndDoCommand (chan, &cmd, 0); + } +} + +static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec) { + + SndCallBackUPP callback; + int sample_bits; + int i; + long initOptions; + + /* Very few conversions are required, but... */ + switch (spec->format) { + case AUDIO_S8: + spec->format = AUDIO_U8; + break; + case AUDIO_U16LSB: + spec->format = AUDIO_S16LSB; + break; + case AUDIO_U16MSB: + spec->format = AUDIO_S16MSB; + break; + } + SDL_CalculateAudioSpec(spec); + + /* initialize bufferCmd header */ + memset (&header, 0, sizeof(header)); + callback = (SndCallBackUPP) NewSndCallBackUPP (callBackProc); + sample_bits = spec->size / spec->samples / spec->channels * 8; + +#ifdef DEBUG_AUDIO + fprintf(stderr, + "Audio format 0x%x, channels = %d, sample_bits = %d, frequency = %d\n", + spec->format, spec->channels, sample_bits, spec->freq); +#endif /* DEBUG_AUDIO */ + + header.numChannels = spec->channels; + header.sampleSize = sample_bits; + header.sampleRate = spec->freq << 16; + header.numFrames = spec->samples; + header.encode = cmpSH; + + /* Note that we install the 16bitLittleEndian Converter if needed. */ + if ( spec->format == 0x8010 ) { + header.compressionID = fixedCompression; + header.format = k16BitLittleEndianFormat; + } + + /* allocate 2 buffers */ + for (i=0; i<2; i++) { + buffer[i] = (UInt8*)malloc (sizeof(UInt8) * spec->size); + if (buffer[i] == NULL) { + SDL_OutOfMemory(); + return (-1); + } + memset (buffer[i], 0, spec->size); + } + + /* Create the sound manager channel */ + channel = (SndChannelPtr)SDL_malloc(sizeof(*channel)); + if ( channel == NULL ) { + SDL_OutOfMemory(); + return(-1); + } + if ( spec->channels >= 2 ) { + initOptions = initStereo; + } else { + initOptions = initMono; + } + channel->userInfo = (long)this; + channel->qLength = 128; + if ( SndNewChannel(&channel, sampledSynth, initOptions, callback) != noErr ) { + SDL_SetError("Unable to create audio channel"); + SDL_free(channel); + channel = NULL; + return(-1); + } + + /* start playback */ + { + SndCommand cmd; + cmd.cmd = callBackCmd; + cmd.param2 = 0; + running = 1; + SndDoCommand (channel, &cmd, 0); + } + + return 1; +} + +static void Mac_CloseAudio(_THIS) { + + int i; + + running = 0; + + if (channel) { + SndDisposeChannel (channel, true); + channel = NULL; + } + + for ( i=0; i<2; ++i ) { + if ( buffer[i] ) { + SDL_free(buffer[i]); + buffer[i] = NULL; + } + } +} + +#else /* !TARGET_API_MAC_CARBON && !USE_RYANS_SOUNDCODE */ + +static void Mac_LockAudio(_THIS) +{ + /* no-op. */ +} + +static void Mac_UnlockAudio(_THIS) +{ + /* no-op. */ +} + + +/* This function is called by Sound Manager when it has exhausted one of + the buffers, so we'll zero it to silence and fill it with audio if + we're not paused. +*/ +static pascal +void sndDoubleBackProc (SndChannelPtr chan, SndDoubleBufferPtr newbuf) +{ + SDL_AudioDevice *audio = (SDL_AudioDevice *)newbuf->dbUserInfo[0]; + + /* If audio is quitting, don't do anything */ + if ( ! audio->enabled ) { + return; + } + memset (newbuf->dbSoundData, 0, audio->spec.size); + newbuf->dbNumFrames = audio->spec.samples; + if ( ! audio->paused ) { + if ( audio->convert.needed ) { + audio->spec.callback(audio->spec.userdata, + (Uint8 *)audio->convert.buf,audio->convert.len); + SDL_ConvertAudio(&audio->convert); +#if 0 + if ( audio->convert.len_cvt != audio->spec.size ) { + /* Uh oh... probably crashes here */; + } +#endif + SDL_memcpy(newbuf->dbSoundData, audio->convert.buf, + audio->convert.len_cvt); + } else { + audio->spec.callback(audio->spec.userdata, + (Uint8 *)newbuf->dbSoundData, audio->spec.size); + } + } + newbuf->dbFlags |= dbBufferReady; +} + +static int DoubleBufferAudio_Available(void) +{ + int available; + NumVersion sndversion; + long response; + + available = 0; + sndversion = SndSoundManagerVersion(); + if ( sndversion.majorRev >= 3 ) { + if ( Gestalt(gestaltSoundAttr, &response) == noErr ) { + if ( (response & (1 << gestaltSndPlayDoubleBuffer)) ) { + available = 1; + } + } + } else { + if ( Gestalt(gestaltSoundAttr, &response) == noErr ) { + if ( (response & (1 << gestaltHasASC)) ) { + available = 1; + } + } + } + return(available); +} + +static void Mac_CloseAudio(_THIS) +{ + int i; + + if ( channel != NULL ) { + /* Clean up the audio channel */ + SndDisposeChannel(channel, true); + channel = NULL; + } + for ( i=0; i<2; ++i ) { + if ( audio_buf[i] ) { + SDL_free(audio_buf[i]); + audio_buf[i] = NULL; + } + } +} + +static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + SndDoubleBufferHeader2 audio_dbh; + int i; + long initOptions; + int sample_bits; + SndDoubleBackUPP doubleBackProc; + + /* Check to make sure double-buffered audio is available */ + if ( ! DoubleBufferAudio_Available() ) { + SDL_SetError("Sound manager doesn't support double-buffering"); + return(-1); + } + + /* Very few conversions are required, but... */ + switch (spec->format) { + case AUDIO_S8: + spec->format = AUDIO_U8; + break; + case AUDIO_U16LSB: + spec->format = AUDIO_S16LSB; + break; + case AUDIO_U16MSB: + spec->format = AUDIO_S16MSB; + break; + } + SDL_CalculateAudioSpec(spec); + + /* initialize the double-back header */ + SDL_memset(&audio_dbh, 0, sizeof(audio_dbh)); + doubleBackProc = NewSndDoubleBackProc (sndDoubleBackProc); + sample_bits = spec->size / spec->samples / spec->channels * 8; + + audio_dbh.dbhNumChannels = spec->channels; + audio_dbh.dbhSampleSize = sample_bits; + audio_dbh.dbhCompressionID = 0; + audio_dbh.dbhPacketSize = 0; + audio_dbh.dbhSampleRate = spec->freq << 16; + audio_dbh.dbhDoubleBack = doubleBackProc; + audio_dbh.dbhFormat = 0; + + /* Note that we install the 16bitLittleEndian Converter if needed. */ + if ( spec->format == 0x8010 ) { + audio_dbh.dbhCompressionID = fixedCompression; + audio_dbh.dbhFormat = k16BitLittleEndianFormat; + } + + /* allocate the 2 double-back buffers */ + for ( i=0; i<2; ++i ) { + audio_buf[i] = SDL_calloc(1, sizeof(SndDoubleBuffer)+spec->size); + if ( audio_buf[i] == NULL ) { + SDL_OutOfMemory(); + return(-1); + } + audio_buf[i]->dbNumFrames = spec->samples; + audio_buf[i]->dbFlags = dbBufferReady; + audio_buf[i]->dbUserInfo[0] = (long)this; + audio_dbh.dbhBufferPtr[i] = audio_buf[i]; + } + + /* Create the sound manager channel */ + channel = (SndChannelPtr)SDL_malloc(sizeof(*channel)); + if ( channel == NULL ) { + SDL_OutOfMemory(); + return(-1); + } + if ( spec->channels >= 2 ) { + initOptions = initStereo; + } else { + initOptions = initMono; + } + channel->userInfo = 0; + channel->qLength = 128; + if ( SndNewChannel(&channel, sampledSynth, initOptions, 0L) != noErr ) { + SDL_SetError("Unable to create audio channel"); + SDL_free(channel); + channel = NULL; + return(-1); + } + + /* Start playback */ + if ( SndPlayDoubleBuffer(channel, (SndDoubleBufferHeaderPtr)&audio_dbh) + != noErr ) { + SDL_SetError("Unable to play double buffered audio"); + return(-1); + } + + return 1; +} + +#endif /* TARGET_API_MAC_CARBON || USE_RYANS_SOUNDCODE */ + diff --git a/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h b/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h new file mode 100644 index 0000000000..90e19c0695 --- /dev/null +++ b/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h @@ -0,0 +1,50 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +#ifndef _SDL_romaudio_h +#define _SDL_romaudio_h + +#include "../SDL_sysaudio.h" + +/* This is Ryan's improved MacOS sound code, with locking support */ +#define USE_RYANS_SOUNDCODE + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* Sound manager audio channel */ + SndChannelPtr channel; +#if defined(TARGET_API_MAC_CARBON) || defined(USE_RYANS_SOUNDCODE) + /* FIXME: Add Ryan's static data here */ +#else + /* Double buffering variables */ + SndDoubleBufferPtr audio_buf[2]; +#endif +}; + +/* Old variable names */ +#define channel (this->hidden->channel) +#define audio_buf (this->hidden->audio_buf) + +#endif /* _SDL_romaudio_h */ diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c new file mode 100644 index 0000000000..46ba690c3e --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c @@ -0,0 +1,215 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + Audio interrupt variables and callback function + + Patrice Mandin +*/ + +#include + +#include +#include +#include +#include + +#include "SDL_audio.h" +#include "SDL_mintaudio.h" +#include "SDL_mintaudio_stfa.h" + +/* The audio device */ + +SDL_AudioDevice *SDL_MintAudio_device; +Uint8 *SDL_MintAudio_audiobuf[2]; /* Pointers to buffers */ +unsigned long SDL_MintAudio_audiosize; /* Length of audio buffer=spec->size */ +volatile unsigned short SDL_MintAudio_numbuf; /* Buffer to play */ +volatile unsigned short SDL_MintAudio_mutex; +volatile unsigned long SDL_MintAudio_clocktics; +cookie_stfa_t *SDL_MintAudio_stfa; +unsigned short SDL_MintAudio_hasfpu; + +/* MiNT thread variables */ +SDL_bool SDL_MintAudio_mint_present; +SDL_bool SDL_MintAudio_quit_thread; +SDL_bool SDL_MintAudio_thread_finished; +long SDL_MintAudio_thread_pid; + +/* The callback function, called by each driver whenever needed */ + +void SDL_MintAudio_Callback(void) +{ + Uint8 *buffer; + SDL_AudioDevice *audio = SDL_MintAudio_device; + + buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + SDL_memset(buffer, audio->spec.silence, audio->spec.size); + + if (audio->paused) + return; + + if (audio->convert.needed) { + int silence; + + if ( audio->convert.src_format == AUDIO_U8 ) { + silence = 0x80; + } else { + silence = 0; + } + SDL_memset(audio->convert.buf, silence, audio->convert.len); + audio->spec.callback(audio->spec.userdata, + (Uint8 *)audio->convert.buf,audio->convert.len); + SDL_ConvertAudio(&audio->convert); + SDL_memcpy(buffer, audio->convert.buf, audio->convert.len_cvt); + } else { + audio->spec.callback(audio->spec.userdata, buffer, audio->spec.size); + } +} + +/* Add a new frequency/clock/predivisor to the current list */ +void SDL_MintAudio_AddFrequency(_THIS, Uint32 frequency, Uint32 clock, + Uint32 prediv, int gpio_bits) +{ + int i, p; + + if (MINTAUDIO_freqcount==MINTAUDIO_maxfreqs) { + return; + } + + /* Search where to insert the frequency (highest first) */ + for (p=0; p MINTAUDIO_frequencies[p].frequency) { + break; + } + } + + /* Put all following ones farer */ + if (MINTAUDIO_freqcount>0) { + for (i=MINTAUDIO_freqcount; i>p; i--) { + SDL_memcpy(&MINTAUDIO_frequencies[i], &MINTAUDIO_frequencies[i-1], sizeof(mint_frequency_t)); + } + } + + /* And insert new one */ + MINTAUDIO_frequencies[p].frequency = frequency; + MINTAUDIO_frequencies[p].masterclock = clock; + MINTAUDIO_frequencies[p].predivisor = prediv; + MINTAUDIO_frequencies[p].gpio_bits = gpio_bits; + + MINTAUDIO_freqcount++; +} + +/* Search for the nearest frequency */ +int SDL_MintAudio_SearchFrequency(_THIS, int desired_freq) +{ + int i; + + /* Only 1 freq ? */ + if (MINTAUDIO_freqcount==1) { + return 0; + } + + /* Check the array */ + for (i=0; i= ((MINTAUDIO_frequencies[i].frequency+ + MINTAUDIO_frequencies[i+1].frequency)>>1)) { + return i; + } + } + + /* Not in the array, give the latest */ + return MINTAUDIO_freqcount-1; +} + +/* Check if FPU is present */ +void SDL_MintAudio_CheckFpu(void) +{ + long cookie_fpu; + + SDL_MintAudio_hasfpu = 0; + if (Getcookie(C__FPU, &cookie_fpu) != C_FOUND) { + return; + } + switch ((cookie_fpu>>16)&0xfffe) { + case 2: + case 4: + case 6: + case 8: + case 16: + SDL_MintAudio_hasfpu = 1; + break; + } +} + +/* The thread function, used under MiNT with xbios */ +int SDL_MintAudio_Thread(long param) +{ + SndBufPtr pointers; + SDL_bool buffers_filled[2] = {SDL_FALSE, SDL_FALSE}; + + SDL_MintAudio_thread_finished = SDL_FALSE; + while (!SDL_MintAudio_quit_thread) { + if (Buffptr(&pointers)!=0) + continue; + + if (( (unsigned long)pointers.play>=(unsigned long)SDL_MintAudio_audiobuf[0]) + && ( (unsigned long)pointers.play<=(unsigned long)SDL_MintAudio_audiobuf[1])) + { + /* DMA is reading buffer #0, setup buffer #1 if not already done */ + if (!buffers_filled[1]) { + SDL_MintAudio_numbuf = 1; + SDL_MintAudio_Callback(); + Setbuffer(0, SDL_MintAudio_audiobuf[1], SDL_MintAudio_audiobuf[1] + SDL_MintAudio_audiosize); + buffers_filled[1]=SDL_TRUE; + buffers_filled[0]=SDL_FALSE; + } + } else { + /* DMA is reading buffer #1, setup buffer #0 if not already done */ + if (!buffers_filled[0]) { + SDL_MintAudio_numbuf = 0; + SDL_MintAudio_Callback(); + Setbuffer(0, SDL_MintAudio_audiobuf[0], SDL_MintAudio_audiobuf[0] + SDL_MintAudio_audiosize); + buffers_filled[0]=SDL_TRUE; + buffers_filled[1]=SDL_FALSE; + } + } + + usleep(100); + } + SDL_MintAudio_thread_finished = SDL_TRUE; + return 0; +} + +void SDL_MintAudio_WaitThread(void) +{ + if (!SDL_MintAudio_mint_present) + return; + + if (SDL_MintAudio_thread_finished) + return; + + SDL_MintAudio_quit_thread = SDL_TRUE; + while (!SDL_MintAudio_thread_finished) { + Syield(); + } +} diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h new file mode 100644 index 0000000000..ba6056ee3a --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h @@ -0,0 +1,121 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MiNT audio driver + + Patrice Mandin +*/ + +#ifndef _SDL_mintaudio_h +#define _SDL_mintaudio_h + +#include "../SDL_sysaudio.h" +#include "SDL_mintaudio_stfa.h" + +/* Hidden "this" pointer for the audio functions */ +#define _THIS SDL_AudioDevice *this + +/* 16 predivisors with 3 clocks max. */ +#define MINTAUDIO_maxfreqs (16*3) + +typedef struct { + Uint32 frequency; + Uint32 masterclock; + Uint32 predivisor; + int gpio_bits; /* in case of external clock */ +} mint_frequency_t; + +struct SDL_PrivateAudioData { + mint_frequency_t frequencies[MINTAUDIO_maxfreqs]; + int freq_count; /* Number of frequencies in the array */ + int numfreq; /* Number of selected frequency */ +}; + +/* Old variable names */ + +#define MINTAUDIO_frequencies (this->hidden->frequencies) +#define MINTAUDIO_freqcount (this->hidden->freq_count) +#define MINTAUDIO_numfreq (this->hidden->numfreq) + +/* _MCH cookie (values>>16) */ +enum { + MCH_ST=0, + MCH_STE, + MCH_TT, + MCH_F30, + MCH_CLONE, + MCH_ARANYM +}; + +/* Master clocks for replay frequencies */ +#define MASTERCLOCK_STE 8010666 /* Not sure of this one */ +#define MASTERCLOCK_TT 16107953 /* Not sure of this one */ +#define MASTERCLOCK_FALCON1 25175000 +#define MASTERCLOCK_FALCON2 32000000 /* Only usable for DSP56K */ +#define MASTERCLOCK_FALCONEXT -1 /* Clock on DSP56K port, unknown */ +#define MASTERCLOCK_44K 22579200 /* Standard clock for 44.1 Khz */ +#define MASTERCLOCK_48K 24576000 /* Standard clock for 48 Khz */ + +/* Master clock predivisors */ +#define MASTERPREDIV_STE 160 +#define MASTERPREDIV_TT 320 +#define MASTERPREDIV_FALCON 256 +#define MASTERPREDIV_MILAN 256 + +/* Variables */ +extern SDL_AudioDevice *SDL_MintAudio_device; +extern Uint8 *SDL_MintAudio_audiobuf[2]; /* Pointers to buffers */ +extern unsigned long SDL_MintAudio_audiosize; /* Length of audio buffer=spec->size */ +extern volatile unsigned short SDL_MintAudio_numbuf; /* Buffer to play */ +extern volatile unsigned short SDL_MintAudio_mutex; +extern cookie_stfa_t *SDL_MintAudio_stfa; +extern volatile unsigned long SDL_MintAudio_clocktics; +extern unsigned short SDL_MintAudio_hasfpu; /* To preserve fpu registers if needed */ + +/* MiNT thread variables */ +extern SDL_bool SDL_MintAudio_mint_present; +extern SDL_bool SDL_MintAudio_quit_thread; +extern SDL_bool SDL_MintAudio_thread_finished; +extern long SDL_MintAudio_thread_pid; + +/* Functions */ +void SDL_MintAudio_Callback(void); +void SDL_MintAudio_AddFrequency(_THIS, Uint32 frequency, Uint32 clock, + Uint32 prediv, int gpio_bits); +int SDL_MintAudio_SearchFrequency(_THIS, int desired_freq); +void SDL_MintAudio_CheckFpu(void); + +/* MiNT thread functions */ +int SDL_MintAudio_Thread(long param); +void SDL_MintAudio_WaitThread(void); + +/* ASM interrupt functions */ +void SDL_MintAudio_GsxbInterrupt(void); +void SDL_MintAudio_EmptyGsxbInterrupt(void); +void SDL_MintAudio_XbiosInterruptMeasureClock(void); +void SDL_MintAudio_XbiosInterrupt(void); +void SDL_MintAudio_Dma8Interrupt(void); +void SDL_MintAudio_StfaInterrupt(void); + +#endif /* _SDL_mintaudio_h */ diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.c new file mode 100644 index 0000000000..61feba3d64 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.c @@ -0,0 +1,357 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MiNT audio driver + using DMA 8bits (hardware access) + + Patrice Mandin +*/ + +/* Mint includes */ +#include +#include +#include + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" + +#include "../../video/ataricommon/SDL_atarimxalloc_c.h" + +#include "SDL_mintaudio.h" +#include "SDL_mintaudio_dma8.h" + +/*--- Defines ---*/ + +#define MINT_AUDIO_DRIVER_NAME "mint_dma8" + +/* Debug print info */ +#define DEBUG_NAME "audio:dma8: " +#if 0 +#define DEBUG_PRINT(what) \ + { \ + printf what; \ + } +#else +#define DEBUG_PRINT(what) +#endif + +/*--- Static variables ---*/ + +static long cookie_snd, cookie_mch; + +/*--- Audio driver functions ---*/ + +static void Mint_CloseAudio(_THIS); +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_LockAudio(_THIS); +static void Mint_UnlockAudio(_THIS); + +/* To check/init hardware audio */ +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec); + +/* Functions called in supervisor mode */ +static void Mint_InitDma(void); +static void Mint_StopReplay(void); +static void Mint_StartReplay(void); + +/*--- Audio driver bootstrap functions ---*/ + +static int Audio_Available(void) +{ + const char *envr = SDL_getenv("SDL_AUDIODRIVER"); + + /* Check if user asked a different audio driver */ + if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) { + DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n")); + return 0; + } + + /* Cookie _MCH present ? if not, assume ST machine */ + if (Getcookie(C__MCH, &cookie_mch) == C_NOTFOUND) { + cookie_mch = MCH_ST; + } + + /* Cookie _SND present ? if not, assume ST machine */ + if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) { + cookie_snd = SND_PSG; + } + + /* Check if we have 8 bits audio */ + if ((cookie_snd & SND_8BIT)==0) { + DEBUG_PRINT((DEBUG_NAME "no 8 bits sound\n")); + return(0); + } + + /* Check if audio is lockable */ + if (cookie_snd & SND_16BIT) { + if (Locksnd()!=1) { + DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n")); + return(0); + } + + Unlocksnd(); + } + + DEBUG_PRINT((DEBUG_NAME "8 bits audio available!\n")); + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Mint_OpenAudio; + this->CloseAudio = Mint_CloseAudio; + this->LockAudio = Mint_LockAudio; + this->UnlockAudio = Mint_UnlockAudio; + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap MINTAUDIO_DMA8_bootstrap = { + MINT_AUDIO_DRIVER_NAME, "MiNT DMA 8 bits audio driver", + Audio_Available, Audio_CreateDevice +}; + +static void Mint_LockAudio(_THIS) +{ + Supexec(Mint_StopReplay); +} + +static void Mint_UnlockAudio(_THIS) +{ + Supexec(Mint_StartReplay); +} + +static void Mint_CloseAudio(_THIS) +{ + Supexec(Mint_StopReplay); + + DEBUG_PRINT((DEBUG_NAME "closeaudio: replay stopped\n")); + + /* Disable interrupt */ + Jdisint(MFP_DMASOUND); + + DEBUG_PRINT((DEBUG_NAME "closeaudio: interrupt disabled\n")); + + /* Wait if currently playing sound */ + while (SDL_MintAudio_mutex != 0) { + } + + DEBUG_PRINT((DEBUG_NAME "closeaudio: no more interrupt running\n")); + + /* Clear buffers */ + if (SDL_MintAudio_audiobuf[0]) { + Mfree(SDL_MintAudio_audiobuf[0]); + SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL; + } + + DEBUG_PRINT((DEBUG_NAME "closeaudio: buffers freed\n")); +} + +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec) +{ + int i, masterprediv, sfreq; + unsigned long masterclock; + + DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + if (spec->channels > 2) + spec->channels = 2; + + /* Check formats available */ + spec->format = AUDIO_S8; + + /* Calculate and select the closest frequency */ + sfreq=0; + masterclock=MASTERCLOCK_STE; + masterprediv=MASTERPREDIV_STE; + switch(cookie_mch>>16) { +/* + case MCH_STE: + masterclock=MASTERCLOCK_STE; + masterprediv=MASTERPREDIV_STE; + break; +*/ + case MCH_TT: + masterclock=MASTERCLOCK_TT; + masterprediv=MASTERPREDIV_TT; + break; + case MCH_F30: + case MCH_ARANYM: + masterclock=MASTERCLOCK_FALCON1; + masterprediv=MASTERPREDIV_FALCON; + sfreq=1; + break; + } + + MINTAUDIO_freqcount=0; + for (i=sfreq;i<4;i++) { + SDL_MintAudio_AddFrequency(this, masterclock/(masterprediv*(1<freq); + spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency; + + DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + return 0; +} + +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + SDL_MintAudio_device = this; + + /* Check audio capabilities */ + if (Mint_CheckAudio(this, spec)==-1) { + return -1; + } + + SDL_CalculateAudioSpec(spec); + + /* Allocate memory for audio buffers in DMA-able RAM */ + DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size)); + + SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM); + if (SDL_MintAudio_audiobuf[0]==NULL) { + SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer"); + return (-1); + } + SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ; + SDL_MintAudio_numbuf=0; + SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2); + SDL_MintAudio_audiosize = spec->size; + SDL_MintAudio_mutex = 0; + + DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0])); + DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1])); + + SDL_MintAudio_CheckFpu(); + + /* Set replay tracks */ + if (cookie_snd & SND_16BIT) { + Settracks(0,0); + Setmontracks(0); + } + + Supexec(Mint_InitDma); + + /* Set interrupt */ + Jdisint(MFP_DMASOUND); + Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_Dma8Interrupt); + Jenabint(MFP_DMASOUND); + + if (cookie_snd & SND_16BIT) { + if (Setinterrupt(SI_TIMERA, SI_PLAY)<0) { + DEBUG_PRINT((DEBUG_NAME "Setinterrupt() failed\n")); + } + } + + Supexec(Mint_StartReplay); + + return(1); /* We don't use threaded audio */ +} + +/* Functions called in supervisor mode */ + +static void Mint_InitDma(void) +{ + unsigned long buffer; + unsigned char mode; + SDL_AudioDevice *this = SDL_MintAudio_device; + + Mint_StopReplay(); + + /* Set buffer */ + buffer = (unsigned long) SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + DMAAUDIO_IO.start_high = (buffer>>16) & 255; + DMAAUDIO_IO.start_mid = (buffer>>8) & 255; + DMAAUDIO_IO.start_low = buffer & 255; + + buffer += SDL_MintAudio_audiosize; + DMAAUDIO_IO.end_high = (buffer>>16) & 255; + DMAAUDIO_IO.end_mid = (buffer>>8) & 255; + DMAAUDIO_IO.end_low = buffer & 255; + + mode = 3-MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor; + if (this->spec.channels==1) { + mode |= 1<<7; + } + DMAAUDIO_IO.sound_ctrl = mode; +} + +static void Mint_StopReplay(void) +{ + /* Stop replay */ + DMAAUDIO_IO.control=0; +} + +static void Mint_StartReplay(void) +{ + /* Start replay */ + DMAAUDIO_IO.control=3; +} diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.h new file mode 100644 index 0000000000..a52e5db7a5 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.h @@ -0,0 +1,85 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + DMA 8bits and Falcon Codec audio definitions + + Patrice Mandin, Didier Méquignon +*/ + +#ifndef _SDL_mintaudio_dma8_h +#define _SDL_mintaudio_dma8_h + +#define DMAAUDIO_IO_BASE (0xffff8900) +struct DMAAUDIO_IO_S { + unsigned char int_ctrl; + unsigned char control; + + unsigned char dummy1; + unsigned char start_high; + unsigned char dummy2; + unsigned char start_mid; + unsigned char dummy3; + unsigned char start_low; + + unsigned char dummy4; + unsigned char cur_high; + unsigned char dummy5; + unsigned char cur_mid; + unsigned char dummy6; + unsigned char cur_low; + + unsigned char dummy7; + unsigned char end_high; + unsigned char dummy8; + unsigned char end_mid; + unsigned char dummy9; + unsigned char end_low; + + unsigned char dummy10[12]; + + unsigned char track_ctrl; /* CODEC only */ + unsigned char sound_ctrl; + unsigned short sound_data; + unsigned short sound_mask; + + unsigned char dummy11[10]; + + unsigned short dev_ctrl; + unsigned short dest_ctrl; + unsigned short sync_div; + unsigned char track_rec; + unsigned char adderin_input; + unsigned char channel_input; + unsigned char channel_amplification; + unsigned char channel_reduction; + + unsigned char dummy12[6]; + + unsigned char data_direction; + unsigned char dummy13; + unsigned char dev_data; +}; +#define DMAAUDIO_IO ((*(volatile struct DMAAUDIO_IO_S *)DMAAUDIO_IO_BASE)) + +#endif /* _SDL_mintaudio_dma8_h */ diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.c new file mode 100644 index 0000000000..8d7716a137 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.c @@ -0,0 +1,436 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MiNT audio driver + using XBIOS functions (GSXB compatible driver) + + Patrice Mandin +*/ + +/* Mint includes */ +#include +#include +#include + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" + +#include "../../video/ataricommon/SDL_atarimxalloc_c.h" + +#include "SDL_mintaudio.h" +#include "SDL_mintaudio_gsxb.h" + +/*--- Defines ---*/ + +#define MINT_AUDIO_DRIVER_NAME "mint_gsxb" + +/* Debug print info */ +#define DEBUG_NAME "audio:gsxb: " +#if 0 +#define DEBUG_PRINT(what) \ + { \ + printf what; \ + } +#else +#define DEBUG_PRINT(what) +#endif + +/*--- Static variables ---*/ + +static long cookie_snd, cookie_gsxb; + +/*--- Audio driver functions ---*/ + +static void Mint_CloseAudio(_THIS); +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_LockAudio(_THIS); +static void Mint_UnlockAudio(_THIS); + +/* To check/init hardware audio */ +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec); + +/* GSXB callbacks */ +static void Mint_GsxbInterrupt(void); +static void Mint_GsxbNullInterrupt(void); + +/*--- Audio driver bootstrap functions ---*/ + +static int Audio_Available(void) +{ + const char *envr = SDL_getenv("SDL_AUDIODRIVER"); + + /* Check if user asked a different audio driver */ + if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) { + DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n")); + return(0); + } + + /* Cookie _SND present ? if not, assume ST machine */ + if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) { + cookie_snd = SND_PSG; + } + + /* Check if we have 16 bits audio */ + if ((cookie_snd & SND_16BIT)==0) { + DEBUG_PRINT((DEBUG_NAME "no 16 bits sound\n")); + return(0); + } + + /* Cookie GSXB present ? */ + cookie_gsxb = (Getcookie(C_GSXB, &cookie_gsxb) == C_FOUND); + + /* Is it GSXB ? */ + if (((cookie_snd & SND_GSXB)==0) || (cookie_gsxb==0)) { + DEBUG_PRINT((DEBUG_NAME "no GSXB audio\n")); + return(0); + } + + /* Check if audio is lockable */ + if (Locksnd()!=1) { + DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n")); + return(0); + } + + Unlocksnd(); + + DEBUG_PRINT((DEBUG_NAME "GSXB audio available!\n")); + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Mint_OpenAudio; + this->CloseAudio = Mint_CloseAudio; + this->LockAudio = Mint_LockAudio; + this->UnlockAudio = Mint_UnlockAudio; + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap MINTAUDIO_GSXB_bootstrap = { + MINT_AUDIO_DRIVER_NAME, "MiNT GSXB audio driver", + Audio_Available, Audio_CreateDevice +}; + +static void Mint_LockAudio(_THIS) +{ + /* Stop replay */ + Buffoper(0); +} + +static void Mint_UnlockAudio(_THIS) +{ + /* Restart replay */ + Buffoper(SB_PLA_ENA|SB_PLA_RPT); +} + +static void Mint_CloseAudio(_THIS) +{ + /* Stop replay */ + Buffoper(0); + + /* Uninstall interrupt */ + if (NSetinterrupt(2, SI_NONE, Mint_GsxbNullInterrupt)<0) { + DEBUG_PRINT((DEBUG_NAME "NSetinterrupt() failed in close\n")); + } + + /* Wait if currently playing sound */ + while (SDL_MintAudio_mutex != 0) { + } + + /* Clear buffers */ + if (SDL_MintAudio_audiobuf[0]) { + Mfree(SDL_MintAudio_audiobuf[0]); + SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL; + } + + /* Unlock sound system */ + Unlocksnd(); +} + +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec) +{ + long snd_format = 0; + int i, resolution, format_signed, format_bigendian; + Uint16 test_format = SDL_FirstAudioFormat(spec->format); + int valid_datatype = 0; + + resolution = spec->format & 0x00ff; + format_signed = ((spec->format & 0x8000)!=0); + format_bigendian = ((spec->format & 0x1000)!=0); + + DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + if (spec->channels > 2) { + spec->channels = 2; /* no more than stereo! */ + } + + while ((!valid_datatype) && (test_format)) { + /* Check formats available */ + snd_format = Sndstatus(SND_QUERYFORMATS); + spec->format = test_format; + resolution = spec->format & 0xff; + format_signed = (spec->format & (1<<15)); + format_bigendian = (spec->format & (1<<12)); + switch (test_format) { + case AUDIO_U8: + case AUDIO_S8: + if (snd_format & SND_FORMAT8) { + valid_datatype = 1; + snd_format = Sndstatus(SND_QUERY8BIT); + } + break; + + case AUDIO_U16LSB: + case AUDIO_S16LSB: + case AUDIO_U16MSB: + case AUDIO_S16MSB: + if (snd_format & SND_FORMAT16) { + valid_datatype = 1; + snd_format = Sndstatus(SND_QUERY16BIT); + } + break; + + default: + test_format = SDL_NextAudioFormat(); + break; + } + } + + if (!valid_datatype) { + SDL_SetError("Unsupported audio format"); + return (-1); + } + + /* Check signed/unsigned format */ + if (format_signed) { + if (snd_format & SND_FORMATSIGNED) { + /* Ok */ + } else if (snd_format & SND_FORMATUNSIGNED) { + /* Give unsigned format */ + spec->format = spec->format & (~0x8000); + } + } else { + if (snd_format & SND_FORMATUNSIGNED) { + /* Ok */ + } else if (snd_format & SND_FORMATSIGNED) { + /* Give signed format */ + spec->format |= 0x8000; + } + } + + if (format_bigendian) { + if (snd_format & SND_FORMATBIGENDIAN) { + /* Ok */ + } else if (snd_format & SND_FORMATLITTLEENDIAN) { + /* Give little endian format */ + spec->format = spec->format & (~0x1000); + } + } else { + if (snd_format & SND_FORMATLITTLEENDIAN) { + /* Ok */ + } else if (snd_format & SND_FORMATBIGENDIAN) { + /* Give big endian format */ + spec->format |= 0x1000; + } + } + + /* Calculate and select the closest frequency */ + MINTAUDIO_freqcount=0; + for (i=1;i<4;i++) { + SDL_MintAudio_AddFrequency(this, + MASTERCLOCK_44K/(MASTERPREDIV_MILAN*(1<freq); + spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency; + + DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + return 0; +} + +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec) +{ + int channels_mode, prediv; + void *buffer; + + /* Stop currently playing sound */ + Buffoper(0); + + /* Set replay tracks */ + Settracks(0,0); + Setmontracks(0); + + /* Select replay format */ + switch (spec->format & 0xff) { + case 8: + if (spec->channels==2) { + channels_mode=STEREO8; + } else { + channels_mode=MONO8; + } + break; + case 16: + if (spec->channels==2) { + channels_mode=STEREO16; + } else { + channels_mode=MONO16; + } + break; + default: + channels_mode=STEREO16; + break; + } + if (Setmode(channels_mode)<0) { + DEBUG_PRINT((DEBUG_NAME "Setmode() failed\n")); + } + + prediv = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor; + Devconnect(DMAPLAY, DAC, CLKEXT, prediv, 1); + + /* Set buffer */ + buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + if (Setbuffer(0, buffer, buffer + spec->size)<0) { + DEBUG_PRINT((DEBUG_NAME "Setbuffer() failed\n")); + } + + /* Install interrupt */ + if (NSetinterrupt(2, SI_PLAY, Mint_GsxbInterrupt)<0) { + DEBUG_PRINT((DEBUG_NAME "NSetinterrupt() failed\n")); + } + + /* Go */ + Buffoper(SB_PLA_ENA|SB_PLA_RPT); + DEBUG_PRINT((DEBUG_NAME "hardware initialized\n")); +} + +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + /* Lock sound system */ + if (Locksnd()!=1) { + SDL_SetError("Mint_OpenAudio: Audio system already in use"); + return(-1); + } + + SDL_MintAudio_device = this; + + /* Check audio capabilities */ + if (Mint_CheckAudio(this, spec)==-1) { + return -1; + } + + SDL_CalculateAudioSpec(spec); + + /* Allocate memory for audio buffers in DMA-able RAM */ + DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size)); + + SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM); + if (SDL_MintAudio_audiobuf[0]==NULL) { + SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer"); + return (-1); + } + SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ; + SDL_MintAudio_numbuf=0; + SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2); + SDL_MintAudio_audiosize = spec->size; + SDL_MintAudio_mutex = 0; + + DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0])); + DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1])); + + SDL_MintAudio_CheckFpu(); + + /* Setup audio hardware */ + Mint_InitAudio(this, spec); + + return(1); /* We don't use threaded audio */ +} + +static void Mint_GsxbInterrupt(void) +{ + Uint8 *newbuf; + + if (SDL_MintAudio_mutex) + return; + + SDL_MintAudio_mutex=1; + + SDL_MintAudio_numbuf ^= 1; + SDL_MintAudio_Callback(); + newbuf = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + Setbuffer(0, newbuf, newbuf + SDL_MintAudio_audiosize); + + SDL_MintAudio_mutex=0; +} + +static void Mint_GsxbNullInterrupt(void) +{ +} diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.h new file mode 100644 index 0000000000..aee26b7ee3 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.h @@ -0,0 +1,104 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + * GSXB audio definitions + * + * Patrice Mandin + */ + +#ifndef _SDL_mintaudio_gsxb_h +#define _SDL_mintaudio_gsxb_h + +#include /* for trap_14_xxx macros */ + +/* Bit 5 in cookie _SND */ + +#define SND_GSXB (1<<5) + +/* NSoundcmd modes */ + +#define SETRATE 7 /* Set sample rate */ +#define SET8BITFORMAT 8 /* 8 bits format */ +#define SET16BITFORMAT 9 /* 16 bits format */ +#define SET24BITFORMAT 10 /* 24 bits format */ +#define SET32BITFORMAT 11 /* 32 bits format */ +#define LTATTEN_MASTER 12 /* Attenuation */ +#define RTATTEN_MASTER 13 +#define LTATTEN_MICIN 14 +#define RTATTEN_MICIN 15 +#define LTATTEN_FMGEN 16 +#define RTATTEN_FMGEN 17 +#define LTATTEN_LINEIN 18 +#define RTATTEN_LINEIN 19 +#define LTATTEN_CDIN 20 +#define RTATTEN_CDIN 21 +#define LTATTEN_VIDIN 22 +#define RTATTEN_VIDIN 23 +#define LTATTEN_AUXIN 24 +#define RTATTEN_AUXIN 25 + +/* Setmode modes */ + +#define MONO16 3 +#define STEREO24 4 +#define STEREO32 5 +#define MONO24 6 +#define MONO32 7 + +/* Sndstatus modes */ + +#define SND_QUERYFORMATS 2 +#define SND_QUERYMIXERS 3 +#define SND_QUERYSOURCES 4 +#define SND_QUERYDUPLEX 5 +#define SND_QUERY8BIT 8 +#define SND_QUERY16BIT 9 +#define SND_QUERY24BIT 10 +#define SND_QUERY32BIT 11 + +#define SND_FORMAT8 (1<<0) +#define SND_FORMAT16 (1<<1) +#define SND_FORMAT24 (1<<2) +#define SND_FORMAT32 (1<<3) + +#define SND_FORMATSIGNED (1<<0) +#define SND_FORMATUNSIGNED (1<<1) +#define SND_FORMATBIGENDIAN (1<<2) +#define SND_FORMATLITTLEENDIAN (1<<3) + +/* Devconnect prescalers */ + +#define CLK_44K 1 +#define CLK_22K 3 +#define CLK_11K 7 + +/* Extra xbios functions */ + +#define NSoundcmd(mode,data,data2) \ + (long)trap_14_wwl((short)130,(short)(mode),(short)(data),(long)(data2)) +#define NSetinterrupt(src_inter,cause,inth_addr) \ + (long)trap_14_wwwl((short)135,(short)(src_inter),(short)(cause), \ + (long)(inth_addr)) + +#endif /* _SDL_mintaudio_gsxb_h */ diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S new file mode 100644 index 0000000000..a2ecac4c65 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S @@ -0,0 +1,386 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ + +/* + Audio interrupts + + Patrice Mandin, Didier Méquignon + */ + + .text + + .globl _SDL_MintAudio_Callback + + .globl _SDL_MintAudio_XbiosInterrupt + .globl _SDL_MintAudio_XbiosInterruptMeasureClock + .globl _SDL_MintAudio_Dma8Interrupt + .globl _SDL_MintAudio_StfaInterrupt + + .globl _SDL_MintAudio_mutex + .globl _SDL_MintAudio_audiobuf + .globl _SDL_MintAudio_numbuf + .globl _SDL_MintAudio_audiosize + .globl _SDL_MintAudio_clocktics + .globl _SDL_MintAudio_hasfpu + + .globl _SDL_MintAudio_stfa + +/* + How it works: + - Audio is playing buffer #0 (resp. #1) + - We must calculate a sample in buffer #1 (resp. #0) + so we first call the callback to do it + - Then we swap the buffers +*/ + +#define savptr 0x4a2 +#define savamt 0x46 + +/*--- Save/restore FPU context ---*/ + +#if defined(__mcoldfire__) + +#define SAVE_FPU_CONTEXT \ + lea sp@(-216),sp; \ + fsave sp@; \ + fmovel fpiar,sp@-; \ + lea sp@(-64),sp; \ + fmovemd fp0-fp7,sp@ + +#define RESTORE_FPU_CONTEXT \ + fmovemd sp@,fp0-fp7; \ + lea sp@(64),sp; \ + fmovel sp@+,fpiar; \ + frestore sp@; \ + lea sp@(216),sp + +#else + +#define SAVE_FPU_CONTEXT \ + .chip 68k/68881; \ + fsave sp@-; \ + fmoveml fpcr/fpsr/fpiar,sp@-; \ + fmovemx fp0-fp7,sp@-; \ + .chip 68k + +#define RESTORE_FPU_CONTEXT \ + .chip 68k/68881; \ + fmovemx sp@+,fp0-fp7; \ + fmoveml sp@+,fpcr/fpsr/fpiar; \ + frestore sp@+; \ + .chip 68k + +#endif + +/*--- Xbios interrupt vector to measure Falcon external clock ---*/ + +_SDL_MintAudio_XbiosInterruptMeasureClock: /* 1 mS */ +#if defined(__mcoldfire__) + movel d0,sp@- + + moveql #0,d0 + btst d0,0xFFFF8901:w /* state DMA sound */ +#else + btst #0,0xFFFF8901:w /* state DMA sound */ +#endif + beqs SDL_MintAudio_EndIntMeasure + addql #1,_SDL_MintAudio_clocktics +SDL_MintAudio_EndIntMeasure: +#if defined(__mcoldfire__) + moveql #5,d0 + bclr d0,0xFFFFFA0F:w /* Clear service bit */ + + movel sp@+,d0 +#else + bclr #5,0xFFFFFA0F:w /* Clear service bit */ +#endif + rte + +/*--- Xbios interrupt vector ---*/ + +_SDL_MintAudio_XbiosInterrupt: +#if defined(__mcoldfire__) + lea sp@(-60),sp + moveml d0-d7/a0-a6,sp@ +#else + moveml d0-d7/a0-a6,sp@- +#endif + + /* Reenable interrupts, so other interrupts can work */ + movew #0x2300,sr + + /* Clear service bit, so other MFP interrupts can work */ +#if defined(__mcoldfire__) + moveql #5,d0 + bclr d0,0xfffffa0f:w +#else + bclr #5,0xfffffa0f:w +#endif + + /* Check if we are not already running */ + tstw _SDL_MintAudio_mutex + bne SDL_MintAudio_XbiosEnd + +#if defined(__mcoldfire__) + movew _SDL_MintAudio_mutex,d0 + notl d0 + movew d0,_SDL_MintAudio_mutex + + movew _SDL_MintAudio_numbuf,d1 + eorl #1,d1 + movew d1,_SDL_MintAudio_numbuf +#else + notw _SDL_MintAudio_mutex + + /* Swap buffers */ + eorw #1,_SDL_MintAudio_numbuf +#endif + + /* Save FPU if needed */ + tstw _SDL_MintAudio_hasfpu + beqs SDL_MintAudio_Xbios_nofpu1 + SAVE_FPU_CONTEXT +SDL_MintAudio_Xbios_nofpu1: + + /* Callback */ + jsr _SDL_MintAudio_Callback + + /* Restore FPU if needed */ + tstw _SDL_MintAudio_hasfpu + beqs SDL_MintAudio_Xbios_nofpu2 + RESTORE_FPU_CONTEXT +SDL_MintAudio_Xbios_nofpu2: + + /* Reserve space for registers */ +#if defined(__mcoldfire__) + movel #savamt,d0 + subl d0,savptr +#else + subl #savamt,savptr +#endif + + /* Set new buffer */ + + moveq #0,d0 + movel _SDL_MintAudio_audiosize,d1 + + movew _SDL_MintAudio_numbuf,d0 + lsll #2,d0 + lea _SDL_MintAudio_audiobuf,a0 + movel a0@(d0:l),a1 + + lea a1@(d1:l),a2 + + movel a2,sp@- + movel a1,sp@- + clrw sp@- + movew #131,sp@- + trap #14 + lea sp@(12),sp + + /* Restore registers space */ +#if defined(__mcoldfire__) + movel #savamt,d0 + addl d0,savptr +#else + addl #savamt,savptr +#endif + + clrw _SDL_MintAudio_mutex +SDL_MintAudio_XbiosEnd: +#if defined(__mcoldfire__) + moveml sp@,d0-d7/a0-a6 + lea sp@(60),sp +#else + moveml sp@+,d0-d7/a0-a6 +#endif + rte + +/*--- DMA 8 bits interrupt vector ---*/ + +_SDL_MintAudio_Dma8Interrupt: +#if defined(__mcoldfire__) + lea sp@(-16),sp + moveml d0-d1/a0-a1,sp@ +#else + moveml d0-d1/a0-a1,sp@- +#endif + + /* Reenable interrupts, so other interrupts can work */ + movew #0x2300,sr + + /* Clear service bit, so other MFP interrupts can work */ +#if defined(__mcoldfire__) + moveql #5,d0 + bclr d0,0xfffffa0f:w +#else + bclr #5,0xfffffa0f:w +#endif + /* Check if we are not already running */ + tstw _SDL_MintAudio_mutex + bne SDL_MintAudio_Dma8End + +#if defined(__mcoldfire__) + movew _SDL_MintAudio_mutex,d0 + notl d0 + movew d0,_SDL_MintAudio_mutex + + movew _SDL_MintAudio_numbuf,d1 + eorl #1,d1 + movew d1,_SDL_MintAudio_numbuf +#else + notw _SDL_MintAudio_mutex + + /* Swap buffers */ + eorw #1,_SDL_MintAudio_numbuf +#endif + + /* Save FPU if needed */ + tstw _SDL_MintAudio_hasfpu + beqs SDL_MintAudio_Dma8_nofpu1 + SAVE_FPU_CONTEXT +SDL_MintAudio_Dma8_nofpu1: + + /* Callback */ + jsr _SDL_MintAudio_Callback + + /* Restore FPU if needed */ + tstw _SDL_MintAudio_hasfpu + beqs SDL_MintAudio_Dma8_nofpu2 + RESTORE_FPU_CONTEXT +SDL_MintAudio_Dma8_nofpu2: + + /* Set new buffer */ + + moveq #0,d0 + + movew _SDL_MintAudio_numbuf,d0 + lsll #2,d0 + lea _SDL_MintAudio_audiobuf,a0 + movel a0@(d0:l),d1 + + /* Modify DMA addresses */ + lea 0xffff8900:w,a0 + + movel d1,d0 + + moveb d0,a0@(0x07) /* Start address */ + lsrl #8,d0 + moveb d0,a0@(0x05) + lsrl #8,d0 + moveb d0,a0@(0x03) + + addl _SDL_MintAudio_audiosize,d1 + + movel d1,d0 + + moveb d0,a0@(0x13) /* End address */ + lsrl #8,d0 + moveb d0,a0@(0x11) + lsrl #8,d0 + moveb d0,a0@(0x0f) + + clrw _SDL_MintAudio_mutex +SDL_MintAudio_Dma8End: +#if defined(__mcoldfire__) + moveml sp@,d0-d1/a0-a1 + lea sp@(16),sp +#else + moveml sp@+,d0-d1/a0-a1 +#endif + rte + +/*--- STFA interrupt vector ---*/ + +STFA_SOUND_START = 6 +STFA_SOUND_END = STFA_SOUND_START+8 + +_SDL_MintAudio_StfaInterrupt: + /* Reenable interrupts, so other interrupts can work */ + movew #0x2300,sr + + /* Check if we are not already running */ + tstw _SDL_MintAudio_mutex + +#if defined(__mcoldfire__) + bne SDL_MintAudio_StfaEnd + + lea sp@(-60),sp + moveml d0-d7/a0-a6,sp@ + + movew _SDL_MintAudio_mutex,d0 + notl d0 + movew d0,_SDL_MintAudio_mutex + + movew _SDL_MintAudio_numbuf,d1 + eorl #1,d1 + movew d1,_SDL_MintAudio_numbuf +#else + bnes SDL_MintAudio_StfaEnd + + moveml d0-d7/a0-a6,sp@- + + notw _SDL_MintAudio_mutex + + /* Swap buffers */ + eorw #1,_SDL_MintAudio_numbuf +#endif + + /* Save FPU if needed */ + tstw _SDL_MintAudio_hasfpu + beqs SDL_MintAudio_Stfa_nofpu1 + SAVE_FPU_CONTEXT +SDL_MintAudio_Stfa_nofpu1: + + /* Callback */ + jsr _SDL_MintAudio_Callback + + /* Restore FPU if needed */ + tstw _SDL_MintAudio_hasfpu + beqs SDL_MintAudio_Stfa_nofpu2 + RESTORE_FPU_CONTEXT +SDL_MintAudio_Stfa_nofpu2: + + /* Set new buffer */ + + moveq #0,d0 + movel _SDL_MintAudio_stfa,a1 + + movew _SDL_MintAudio_numbuf,d0 + lsll #2,d0 + lea _SDL_MintAudio_audiobuf,a0 + movel a0@(d0:l),d1 + + /* Modify STFA replay buffers */ + movel d1,a1@(STFA_SOUND_START) + addl _SDL_MintAudio_audiosize,d1 + movel d1,a1@(STFA_SOUND_END) + +#if defined(__mcoldfire__) + moveml sp@,d0-d7/a0-a6 + lea sp@(60),sp +#else + moveml sp@+,d0-d7/a0-a6 +#endif + clrw _SDL_MintAudio_mutex +SDL_MintAudio_StfaEnd: + rte diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.c new file mode 100644 index 0000000000..387609b168 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.c @@ -0,0 +1,405 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MiNT audio driver + using XBIOS functions (MacSound compatible driver) + + Patrice Mandin +*/ + +#include + +/* Mint includes */ +#include +#include +#include + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" + +#include "../../video/ataricommon/SDL_atarimxalloc_c.h" + +#include "SDL_mintaudio.h" +#include "SDL_mintaudio_mcsn.h" + +/*--- Defines ---*/ + +#define MINT_AUDIO_DRIVER_NAME "mint_mcsn" + +/* Debug print info */ +#define DEBUG_NAME "audio:mcsn: " +#if 0 +#define DEBUG_PRINT(what) \ + { \ + printf what; \ + } +#else +#define DEBUG_PRINT(what) +#endif + +/*--- Static variables ---*/ + +static long cookie_snd, cookie_mch; +static cookie_mcsn_t *cookie_mcsn; + +/*--- Audio driver functions ---*/ + +static void Mint_CloseAudio(_THIS); +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_LockAudio(_THIS); +static void Mint_UnlockAudio(_THIS); + +/* To check/init hardware audio */ +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec); + +/*--- Audio driver bootstrap functions ---*/ + +static int Audio_Available(void) +{ + long dummy; + const char *envr = SDL_getenv("SDL_AUDIODRIVER"); + + SDL_MintAudio_mint_present = (Getcookie(C_MiNT, &dummy) == C_FOUND); + + /* We can't use XBIOS in interrupt with Magic, don't know about thread */ + if (Getcookie(C_MagX, &dummy) == C_FOUND) { + return(0); + } + + /* Check if user asked a different audio driver */ + if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) { + DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n")); + return(0); + } + + /* Cookie _MCH present ? if not, assume ST machine */ + if (Getcookie(C__MCH, &cookie_mch) == C_NOTFOUND) { + cookie_mch = MCH_ST; + } + + /* Cookie _SND present ? if not, assume ST machine */ + if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) { + cookie_snd = SND_PSG; + } + + /* Check if we have 16 bits audio */ + if ((cookie_snd & SND_16BIT)==0) { + DEBUG_PRINT((DEBUG_NAME "no 16 bits sound\n")); + return(0); + } + + /* Cookie MCSN present ? */ + if (Getcookie(C_McSn, &dummy) != C_FOUND) { + DEBUG_PRINT((DEBUG_NAME "no MCSN audio\n")); + return(0); + } + cookie_mcsn = (cookie_mcsn_t *) dummy; + + /* Check if interrupt at end of replay */ + if (cookie_mcsn->pint == 0) { + DEBUG_PRINT((DEBUG_NAME "no interrupt at end of replay\n")); + return(0); + } + + /* Check if audio is lockable */ + if (Locksnd()!=1) { + DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n")); + return(0); + } + + Unlocksnd(); + + DEBUG_PRINT((DEBUG_NAME "MCSN audio available!\n")); + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Mint_OpenAudio; + this->CloseAudio = Mint_CloseAudio; + this->LockAudio = Mint_LockAudio; + this->UnlockAudio = Mint_UnlockAudio; + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap MINTAUDIO_MCSN_bootstrap = { + MINT_AUDIO_DRIVER_NAME, "MiNT MCSN audio driver", + Audio_Available, Audio_CreateDevice +}; + +static void Mint_LockAudio(_THIS) +{ + /* Stop replay */ + Buffoper(0); +} + +static void Mint_UnlockAudio(_THIS) +{ + /* Restart replay */ + Buffoper(SB_PLA_ENA|SB_PLA_RPT); +} + +static void Mint_CloseAudio(_THIS) +{ + /* Stop replay */ + SDL_MintAudio_WaitThread(); + Buffoper(0); + + if (!SDL_MintAudio_mint_present) { + /* Uninstall interrupt */ + Jdisint(MFP_DMASOUND); + } + + /* Wait if currently playing sound */ + while (SDL_MintAudio_mutex != 0) { + } + + /* Clear buffers */ + if (SDL_MintAudio_audiobuf[0]) { + Mfree(SDL_MintAudio_audiobuf[0]); + SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL; + } + + /* Unlock sound system */ + Unlocksnd(); +} + +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec) +{ + int i; + unsigned long masterclock, masterprediv; + + DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + if (spec->channels > 2) { + spec->channels = 2; /* no more than stereo! */ + } + + /* Check formats available */ + MINTAUDIO_freqcount=0; + switch(cookie_mcsn->play) { + case MCSN_ST: + spec->channels=1; + spec->format=8; /* FIXME: is it signed or unsigned ? */ + SDL_MintAudio_AddFrequency(this, 12500, 0, 0, -1); + break; + case MCSN_TT: /* Also STE, Mega STE */ + spec->format=AUDIO_S8; + masterclock=MASTERCLOCK_STE; + masterprediv=MASTERPREDIV_STE; + if ((cookie_mch>>16)==MCH_TT) { + masterclock=MASTERCLOCK_TT; + masterprediv=MASTERPREDIV_TT; + } + for (i=0; i<4; i++) { + SDL_MintAudio_AddFrequency(this, masterclock/(masterprediv*(1<res1 != 0) { + for (i=1; i<4; i++) { + SDL_MintAudio_AddFrequency(this, (cookie_mcsn->res1)/(MASTERPREDIV_FALCON*(1<format |= 0x8000; /* Audio is always signed */ + if ((spec->format & 0x00ff)==16) { + spec->format |= 0x1000; /* Audio is always big endian */ + spec->channels=2; /* 16 bits always stereo */ + } + break; + } + +#if 0 + for (i=0; ifreq); + spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency; + + DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + return 0; +} + +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec) +{ + int channels_mode, prediv, dmaclock; + void *buffer; + + /* Stop currently playing sound */ + SDL_MintAudio_quit_thread = SDL_FALSE; + SDL_MintAudio_thread_finished = SDL_TRUE; + SDL_MintAudio_WaitThread(); + Buffoper(0); + + /* Set replay tracks */ + Settracks(0,0); + Setmontracks(0); + + /* Select replay format */ + channels_mode=STEREO16; + switch (spec->format & 0xff) { + case 8: + if (spec->channels==2) { + channels_mode=STEREO8; + } else { + channels_mode=MONO8; + } + break; + } + if (Setmode(channels_mode)<0) { + DEBUG_PRINT((DEBUG_NAME "Setmode() failed\n")); + } + + dmaclock = MINTAUDIO_frequencies[MINTAUDIO_numfreq].masterclock; + prediv = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor; + switch(cookie_mcsn->play) { + case MCSN_TT: + Devconnect(DMAPLAY, DAC, CLK25M, CLKOLD, 1); + Soundcmd(SETPRESCALE, prediv); + DEBUG_PRINT((DEBUG_NAME "STE/TT prescaler selected\n")); + break; + case MCSN_FALCON: + Devconnect(DMAPLAY, DAC, dmaclock, prediv, 1); + DEBUG_PRINT((DEBUG_NAME "Falcon prescaler selected\n")); + break; + } + + /* Set buffer */ + buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + if (Setbuffer(0, buffer, buffer + spec->size)<0) { + DEBUG_PRINT((DEBUG_NAME "Setbuffer() failed\n")); + } + + if (SDL_MintAudio_mint_present) { + SDL_MintAudio_thread_pid = tfork(SDL_MintAudio_Thread, 0); + } else { + /* Install interrupt */ + Jdisint(MFP_DMASOUND); + Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_XbiosInterrupt); + Jenabint(MFP_DMASOUND); + + if (Setinterrupt(SI_TIMERA, SI_PLAY)<0) { + DEBUG_PRINT((DEBUG_NAME "Setinterrupt() failed\n")); + } + } + + /* Go */ + Buffoper(SB_PLA_ENA|SB_PLA_RPT); + DEBUG_PRINT((DEBUG_NAME "hardware initialized\n")); +} + +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + /* Lock sound system */ + if (Locksnd()!=1) { + SDL_SetError("Mint_OpenAudio: Audio system already in use"); + return(-1); + } + + SDL_MintAudio_device = this; + + /* Check audio capabilities */ + if (Mint_CheckAudio(this, spec)==-1) { + return -1; + } + + SDL_CalculateAudioSpec(spec); + + /* Allocate memory for audio buffers in DMA-able RAM */ + DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size)); + + SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM); + if (SDL_MintAudio_audiobuf[0]==NULL) { + SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer"); + return (-1); + } + SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ; + SDL_MintAudio_numbuf=0; + SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2); + SDL_MintAudio_audiosize = spec->size; + SDL_MintAudio_mutex = 0; + + DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0])); + DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1])); + + SDL_MintAudio_CheckFpu(); + + /* Setup audio hardware */ + Mint_InitAudio(this, spec); + + return(1); /* We don't use SDL threaded audio */ +} diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.h new file mode 100644 index 0000000000..b772fdab03 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.h @@ -0,0 +1,59 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MCSN control structure + + Patrice Mandin +*/ + +#ifndef _SDL_mintaudio_mcsh_h +#define _SDL_mintaudio_mcsh_h + +typedef struct { + unsigned short version; /* Version */ + unsigned short size; /* Size of structure */ + + unsigned short play; /* Replay capability */ + unsigned short record; /* Record capability */ + unsigned short dsp; /* DSP56K present */ + unsigned short pint; /* Interrupt at end of replay */ + unsigned short rint; /* Interrupt at end of record */ + + unsigned long res1; /* Frequency of external clock */ + unsigned long res2; + unsigned long res3; + unsigned long res4; +} cookie_mcsn_t; + +enum { + MCSN_ST=0, + MCSN_TT, + MCSN_STE=MCSN_TT, + MCSN_FALCON, + MCSN_MAC=MCSN_FALCON +}; + +#define SETSMPFREQ 7 /* Set sample frequency */ + +#endif /* _SDL_mintaudio_mcsh_h */ diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.c new file mode 100644 index 0000000000..4a581e0351 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.c @@ -0,0 +1,326 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MiNT audio driver + using XBIOS functions (STFA driver) + + Patrice Mandin +*/ + +/* Mint includes */ +#include +#include +#include + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" + +#include "../../video/ataricommon/SDL_atarimxalloc_c.h" +#include "../../video/ataricommon/SDL_atarisuper.h" + +#include "SDL_mintaudio.h" +#include "SDL_mintaudio_stfa.h" + +/*--- Defines ---*/ + +#define MINT_AUDIO_DRIVER_NAME "mint_stfa" + +/* Debug print info */ +#define DEBUG_NAME "audio:stfa: " +#if 0 +#define DEBUG_PRINT(what) \ + { \ + printf what; \ + } +#else +#define DEBUG_PRINT(what) +#endif + +/*--- Static variables ---*/ + +static long cookie_snd, cookie_mch; +static cookie_stfa_t *cookie_stfa; + +static const int freqs[16]={ + 4995, 6269, 7493, 8192, + 9830, 10971, 12538, 14985, + 16384, 19819, 21943, 24576, + 30720, 32336, 43885, 49152 +}; + +/*--- Audio driver functions ---*/ + +static void Mint_CloseAudio(_THIS); +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_LockAudio(_THIS); +static void Mint_UnlockAudio(_THIS); + +/* To check/init hardware audio */ +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec); + +/*--- Audio driver bootstrap functions ---*/ + +static int Audio_Available(void) +{ + long dummy; + const char *envr = SDL_getenv("SDL_AUDIODRIVER"); + + /* Check if user asked a different audio driver */ + if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) { + DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n")); + return(0); + } + + /* Cookie _MCH present ? if not, assume ST machine */ + if (Getcookie(C__MCH, &cookie_mch) == C_NOTFOUND) { + cookie_mch = MCH_ST; + } + + /* Cookie _SND present ? if not, assume ST machine */ + if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) { + cookie_snd = SND_PSG; + } + + /* Cookie STFA present ? */ + if (Getcookie(C_STFA, &dummy) != C_FOUND) { + DEBUG_PRINT((DEBUG_NAME "no STFA audio\n")); + return(0); + } + cookie_stfa = (cookie_stfa_t *) dummy; + + SDL_MintAudio_stfa = cookie_stfa; + + DEBUG_PRINT((DEBUG_NAME "STFA audio available!\n")); + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Mint_OpenAudio; + this->CloseAudio = Mint_CloseAudio; + this->LockAudio = Mint_LockAudio; + this->UnlockAudio = Mint_UnlockAudio; + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap MINTAUDIO_STFA_bootstrap = { + MINT_AUDIO_DRIVER_NAME, "MiNT STFA audio driver", + Audio_Available, Audio_CreateDevice +}; + +static void Mint_LockAudio(_THIS) +{ + void *oldpile; + + /* Stop replay */ + oldpile=(void *)Super(0); + cookie_stfa->sound_enable=STFA_PLAY_DISABLE; + SuperToUser(oldpile); +} + +static void Mint_UnlockAudio(_THIS) +{ + void *oldpile; + + /* Restart replay */ + oldpile=(void *)Super(0); + cookie_stfa->sound_enable=STFA_PLAY_ENABLE|STFA_PLAY_REPEAT; + SuperToUser(oldpile); +} + +static void Mint_CloseAudio(_THIS) +{ + void *oldpile; + + /* Stop replay */ + oldpile=(void *)Super(0); + cookie_stfa->sound_enable=STFA_PLAY_DISABLE; + SuperToUser(oldpile); + + /* Wait if currently playing sound */ + while (SDL_MintAudio_mutex != 0) { + } + + /* Clear buffers */ + if (SDL_MintAudio_audiobuf[0]) { + Mfree(SDL_MintAudio_audiobuf[0]); + SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL; + } +} + +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec) +{ + int i; + + DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + if (spec->channels > 2) { + spec->channels = 2; /* no more than stereo! */ + } + + /* Check formats available */ + MINTAUDIO_freqcount=0; + for (i=0;i<16;i++) { + SDL_MintAudio_AddFrequency(this, freqs[i], 0, i, -1); + } + +#if 1 + for (i=0; ifreq); + spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency; + + DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + return 0; +} + +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec) +{ + void *buffer; + void *oldpile; + + buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + + oldpile=(void *)Super(0); + + /* Stop replay */ + cookie_stfa->sound_enable=STFA_PLAY_DISABLE; + + /* Select replay format */ + cookie_stfa->sound_control = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor; + if ((spec->format & 0xff)==8) { + cookie_stfa->sound_control |= STFA_FORMAT_8BIT; + } else { + cookie_stfa->sound_control |= STFA_FORMAT_16BIT; + } + if (spec->channels==2) { + cookie_stfa->sound_control |= STFA_FORMAT_STEREO; + } else { + cookie_stfa->sound_control |= STFA_FORMAT_MONO; + } + if ((spec->format & 0x8000)!=0) { + cookie_stfa->sound_control |= STFA_FORMAT_SIGNED; + } else { + cookie_stfa->sound_control |= STFA_FORMAT_UNSIGNED; + } + if ((spec->format & 0x1000)!=0) { + cookie_stfa->sound_control |= STFA_FORMAT_BIGENDIAN; + } else { + cookie_stfa->sound_control |= STFA_FORMAT_LITENDIAN; + } + + /* Set buffer */ + cookie_stfa->sound_start = (unsigned long) buffer; + cookie_stfa->sound_end = (unsigned long) (buffer + spec->size); + + /* Set interrupt */ + cookie_stfa->stfa_it = SDL_MintAudio_StfaInterrupt; + + /* Restart replay */ + cookie_stfa->sound_enable=STFA_PLAY_ENABLE|STFA_PLAY_REPEAT; + + SuperToUser(oldpile); + + DEBUG_PRINT((DEBUG_NAME "hardware initialized\n")); +} + +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + SDL_MintAudio_device = this; + + /* Check audio capabilities */ + if (Mint_CheckAudio(this, spec)==-1) { + return -1; + } + + SDL_CalculateAudioSpec(spec); + + /* Allocate memory for audio buffers in DMA-able RAM */ + DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size)); + + SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM); + if (SDL_MintAudio_audiobuf[0]==NULL) { + SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer"); + return (-1); + } + SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ; + SDL_MintAudio_numbuf=0; + SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2); + SDL_MintAudio_audiosize = spec->size; + SDL_MintAudio_mutex = 0; + + DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0])); + DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1])); + + SDL_MintAudio_CheckFpu(); + + /* Setup audio hardware */ + Mint_InitAudio(this, spec); + + return(1); /* We don't use threaded audio */ +} diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.h new file mode 100644 index 0000000000..1789b4bb41 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.h @@ -0,0 +1,97 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + STFA control structure + + Patrice Mandin +*/ + +#ifndef _SDL_mintaudio_stfa_h +#define _SDL_mintaudio_stfa_h + +/*--- Defines ---*/ + +#define STFA_PLAY_ENABLE (1<<0) +#define STFA_PLAY_DISABLE (0<<0) +#define STFA_PLAY_REPEAT (1<<1) +#define STFA_PLAY_SINGLE (0<<1) + +#define STFA_FORMAT_SIGNED (1<<15) +#define STFA_FORMAT_UNSIGNED (0<<15) +#define STFA_FORMAT_STEREO (1<<14) +#define STFA_FORMAT_MONO (0<<14) +#define STFA_FORMAT_16BIT (1<<13) +#define STFA_FORMAT_8BIT (0<<13) +#define STFA_FORMAT_LITENDIAN (1<<9) +#define STFA_FORMAT_BIGENDIAN (0<<9) +#define STFA_FORMAT_FREQ_MASK 0x0f +enum { + STFA_FORMAT_F4995=0, + STFA_FORMAT_F6269, + STFA_FORMAT_F7493, + STFA_FORMAT_F8192, + + STFA_FORMAT_F9830, + STFA_FORMAT_F10971, + STFA_FORMAT_F12538, + STFA_FORMAT_F14985, + + STFA_FORMAT_F16384, + STFA_FORMAT_F19819, + STFA_FORMAT_F21943, + STFA_FORMAT_F24576, + + STFA_FORMAT_F30720, + STFA_FORMAT_F32336, + STFA_FORMAT_F43885, + STFA_FORMAT_F49152 +}; + +/*--- Types ---*/ + +typedef struct { + unsigned short sound_enable; + unsigned short sound_control; + unsigned short sound_output; + unsigned long sound_start; + unsigned long sound_current; + unsigned long sound_end; + unsigned short version; + void *old_vbl; + void *old_timera; + unsigned long old_mfp_status; + void *new_vbl; + void *drivers_list; + void *play_stop; + unsigned short frequency; + void *set_frequency; + unsigned short frequency_threshold; + unsigned short *custom_freq_table; + unsigned short stfa_on_off; + void *new_drivers_list; + unsigned long old_bit_2_of_cookie_snd; + void (*stfa_it)(void); +} cookie_stfa_t; + +#endif /* _SDL_mintaudio_stfa_h */ diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_xbios.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_xbios.c new file mode 100644 index 0000000000..42a0d4a2f9 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_xbios.c @@ -0,0 +1,490 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* + MiNT audio driver + using XBIOS functions (Falcon) + + Patrice Mandin, Didier Méquignon +*/ + +#include +#include + +/* Mint includes */ +#include +#include +#include + +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "../SDL_sysaudio.h" + +#include "../../video/ataricommon/SDL_atarimxalloc_c.h" +#include "../../video/ataricommon/SDL_atarisuper.h" + +#include "SDL_mintaudio.h" +#include "SDL_mintaudio_dma8.h" + +/*--- Defines ---*/ + +#define MINT_AUDIO_DRIVER_NAME "mint_xbios" + +/* Debug print info */ +#define DEBUG_NAME "audio:xbios: " +#if 0 +#define DEBUG_PRINT(what) \ + { \ + printf what; \ + } +#else +#define DEBUG_PRINT(what) +#endif + +/*--- Static variables ---*/ + +static long cookie_snd; + +/*--- Audio driver functions ---*/ + +static void Mint_CloseAudio(_THIS); +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_LockAudio(_THIS); +static void Mint_UnlockAudio(_THIS); + +/* To check/init hardware audio */ +static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec); +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec); + +/*--- Audio driver bootstrap functions ---*/ + +static int Audio_Available(void) +{ +/* unsigned long dummy;*/ + const char *envr = SDL_getenv("SDL_AUDIODRIVER"); + + /*SDL_MintAudio_mint_present = (Getcookie(C_MiNT, &dummy) == C_FOUND);*/ + SDL_MintAudio_mint_present = SDL_FALSE; + + /* We can't use XBIOS in interrupt with Magic, don't know about thread */ + /*if (Getcookie(C_MagX, &dummy) == C_FOUND) { + return(0); + }*/ + + /* Check if user asked a different audio driver */ + if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) { + DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n")); + return(0); + } + + /* Cookie _SND present ? if not, assume ST machine */ + if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) { + cookie_snd = SND_PSG; + } + + /* Check if we have 16 bits audio */ + if ((cookie_snd & SND_16BIT)==0) { + DEBUG_PRINT((DEBUG_NAME "no 16 bits sound\n")); + return(0); + } + + /* Check if audio is lockable */ + if (Locksnd()!=1) { + DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n")); + return(0); + } + + Unlocksnd(); + + DEBUG_PRINT((DEBUG_NAME "XBIOS audio available!\n")); + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = Mint_OpenAudio; + this->CloseAudio = Mint_CloseAudio; + this->LockAudio = Mint_LockAudio; + this->UnlockAudio = Mint_UnlockAudio; + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap MINTAUDIO_XBIOS_bootstrap = { + MINT_AUDIO_DRIVER_NAME, "MiNT XBIOS audio driver", + Audio_Available, Audio_CreateDevice +}; + +static void Mint_LockAudio(_THIS) +{ + /* Stop replay */ + Buffoper(0); +} + +static void Mint_UnlockAudio(_THIS) +{ + /* Restart replay */ + Buffoper(SB_PLA_ENA|SB_PLA_RPT); +} + +static void Mint_CloseAudio(_THIS) +{ + /* Stop replay */ + SDL_MintAudio_WaitThread(); + Buffoper(0); + + if (!SDL_MintAudio_mint_present) { + /* Uninstall interrupt */ + Jdisint(MFP_DMASOUND); + } + + /* Wait if currently playing sound */ + while (SDL_MintAudio_mutex != 0) { + } + + /* Clear buffers */ + if (SDL_MintAudio_audiobuf[0]) { + Mfree(SDL_MintAudio_audiobuf[0]); + SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL; + } + + /* Unlock sound system */ + Unlocksnd(); +} + +/* Falcon XBIOS implementation of Devconnect() is buggy with external clock */ +static void Devconnect2(int src, int dst, int sclk, int pre) +{ + static const unsigned short MASK1[3] = { 0, 0x6000, 0 }; + static const unsigned short MASK2[4] = { 0xFFF0, 0xFF8F, 0xF0FF, 0x0FFF }; + static const unsigned short INDEX1[4] = { 1, 3, 5, 7 }; + static const unsigned short INDEX2[4] = { 0, 2, 4, 6 }; + unsigned short sync_div,dev_ctrl,dest_ctrl; + void *oldstack; + + if (dst==0) { + return; + } + + oldstack=(void *)Super(0); + + dev_ctrl = DMAAUDIO_IO.dev_ctrl; + dest_ctrl = DMAAUDIO_IO.dest_ctrl; + dev_ctrl &= MASK2[src]; + + if (src==ADC) { + dev_ctrl |= MASK1[sclk]; + } else { + dev_ctrl |= (INDEX1[sclk] << (src<<4)); + } + + if (dst & DMAREC) { + dest_ctrl &= 0xFFF0; + dest_ctrl |= INDEX1[src]; + } + + if (dst & DSPRECV) { + dest_ctrl &= 0xFF8F; + dest_ctrl |= (INDEX1[src]<<4); + } + + if (dst & EXTOUT) { + dest_ctrl &= 0xF0FF; + dest_ctrl |= (INDEX1[src]<<8); + } + + if (dst & DAC) { + dev_ctrl &= 0x0FFF; + dev_ctrl |= MASK1[sclk]; + dest_ctrl &= 0x0FFF; + dest_ctrl |= (INDEX2[src]<<12); + } + + sync_div = DMAAUDIO_IO.sync_div; + if (sclk==CLKEXT) { + pre<<=8; + sync_div &= 0xF0FF; + } else { + sync_div &= 0xFFF0; + } + sync_div |= pre; + + DMAAUDIO_IO.dev_ctrl = dev_ctrl; + DMAAUDIO_IO.dest_ctrl = dest_ctrl; + DMAAUDIO_IO.sync_div = sync_div; + + SuperToUser(oldstack); +} + +static void Mint_CheckExternalClock(_THIS) +{ +#define SIZE_BUF_CLOCK_MEASURE (44100/10) + + char *buffer; + int i, j; + + /* DSP present with its GPIO port ? */ + if ((cookie_snd & SND_DSP)==0) { + return; + } + + buffer = Atari_SysMalloc(SIZE_BUF_CLOCK_MEASURE, MX_STRAM); + if (buffer==NULL) { + DEBUG_PRINT((DEBUG_NAME "Not enough memory for the measure\n")); + return; + } + SDL_memset(buffer, 0, SIZE_BUF_CLOCK_MEASURE); + + Buffoper(0); + Settracks(0,0); + Setmontracks(0); + Setmode(MONO8); + Jdisint(MFP_TIMERA); + + for (i=0; i<2; i++) { + Gpio(GPIO_SET,7); /* DSP port gpio outputs */ + Gpio(GPIO_WRITE,2+i); /* 22.5792/24.576 MHz for 44.1/48KHz */ + Devconnect2(DMAPLAY, DAC, CLKEXT, CLK50K); /* Matrix and clock source */ + Setbuffer(0, buffer, buffer + SIZE_BUF_CLOCK_MEASURE); /* Set buffer */ + Xbtimer(XB_TIMERA, 5, 38, SDL_MintAudio_XbiosInterruptMeasureClock); /* delay mode timer A, prediv /64, 1KHz */ + Jenabint(MFP_TIMERA); + SDL_MintAudio_clocktics = 0; + Buffoper(SB_PLA_ENA); + usleep(110000); + + if((Buffoper(-1) & 1)==0) { + if (SDL_MintAudio_clocktics) { + unsigned long khz; + + khz = ((SIZE_BUF_CLOCK_MEASURE/SDL_MintAudio_clocktics) +1) & 0xFFFFFFFE; + DEBUG_PRINT((DEBUG_NAME "measure %d: freq=%lu KHz\n", i+1, khz)); + + if(khz==44) { + for (j=1; j<4; j++) { + SDL_MintAudio_AddFrequency(this, MASTERCLOCK_44K/(MASTERPREDIV_FALCON*(1<format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + if (spec->channels > 2) { + spec->channels = 2; /* no more than stereo! */ + } + + spec->format |= 0x8000; /* Audio is always signed */ + if ((spec->format & 0x00ff)==16) { + spec->format |= 0x1000; /* Audio is always big endian */ + spec->channels=2; /* 16 bits always stereo */ + } + + MINTAUDIO_freqcount=0; + + /* Add external clocks if present */ + Mint_CheckExternalClock(this); + + /* Standard clocks */ + for (i=1;i<12;i++) { + /* Remove unusable Falcon codec predivisors */ + if ((i==6) || (i==8) || (i==10)) { + continue; + } + SDL_MintAudio_AddFrequency(this, MASTERCLOCK_FALCON1/(MASTERPREDIV_FALCON*(i+1)), MASTERCLOCK_FALCON1, i, -1); + } + +#if 1 + for (i=0; ifreq); + spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency; + + DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff)); + DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0))); + DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0))); + DEBUG_PRINT(("channels=%d, ", spec->channels)); + DEBUG_PRINT(("freq=%d\n", spec->freq)); + + return 0; +} + +static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec) +{ + int channels_mode, prediv; + void *buffer; + + /* Stop currently playing sound */ + SDL_MintAudio_quit_thread = SDL_FALSE; + SDL_MintAudio_thread_finished = SDL_TRUE; + SDL_MintAudio_WaitThread(); + Buffoper(0); + + /* Set replay tracks */ + Settracks(0,0); + Setmontracks(0); + + /* Select replay format */ + channels_mode=STEREO16; + switch (spec->format & 0xff) { + case 8: + if (spec->channels==2) { + channels_mode=STEREO8; + } else { + channels_mode=MONO8; + } + break; + } + if (Setmode(channels_mode)<0) { + DEBUG_PRINT((DEBUG_NAME "Setmode() failed\n")); + } + + prediv = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor; + if (MINTAUDIO_frequencies[MINTAUDIO_numfreq].gpio_bits != -1) { + Gpio(GPIO_SET,7); /* DSP port gpio outputs */ + Gpio(GPIO_WRITE, MINTAUDIO_frequencies[MINTAUDIO_numfreq].gpio_bits); + Devconnect2(DMAPLAY, DAC|EXTOUT, CLKEXT, prediv); + } else { + Devconnect2(DMAPLAY, DAC, CLK25M, prediv); + } + + /* Set buffer */ + buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf]; + if (Setbuffer(0, buffer, buffer + spec->size)<0) { + DEBUG_PRINT((DEBUG_NAME "Setbuffer() failed\n")); + } + + if (SDL_MintAudio_mint_present) { + SDL_MintAudio_thread_pid = tfork(SDL_MintAudio_Thread, 0); + } else { + /* Install interrupt */ + Jdisint(MFP_DMASOUND); + /*Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_XbiosInterrupt);*/ + Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_Dma8Interrupt); + Jenabint(MFP_DMASOUND); + + if (Setinterrupt(SI_TIMERA, SI_PLAY)<0) { + DEBUG_PRINT((DEBUG_NAME "Setinterrupt() failed\n")); + } + } + + /* Go */ + Buffoper(SB_PLA_ENA|SB_PLA_RPT); + DEBUG_PRINT((DEBUG_NAME "hardware initialized\n")); +} + +static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + /* Lock sound system */ + if (Locksnd()!=1) { + SDL_SetError("Mint_OpenAudio: Audio system already in use"); + return(-1); + } + + SDL_MintAudio_device = this; + + /* Check audio capabilities */ + if (Mint_CheckAudio(this, spec)==-1) { + return -1; + } + + SDL_CalculateAudioSpec(spec); + + /* Allocate memory for audio buffers in DMA-able RAM */ + DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size)); + + SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM); + if (SDL_MintAudio_audiobuf[0]==NULL) { + SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer"); + return (-1); + } + SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ; + SDL_MintAudio_numbuf=0; + SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2); + SDL_MintAudio_audiosize = spec->size; + SDL_MintAudio_mutex = 0; + + DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0])); + DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1])); + + SDL_MintAudio_CheckFpu(); + + /* Setup audio hardware */ + Mint_InitAudio(this, spec); + + return(1); /* We don't use SDL threaded audio */ +} diff --git a/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c b/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c new file mode 100644 index 0000000000..64a0ecc36d --- /dev/null +++ b/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c @@ -0,0 +1,264 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Tru64 UNIX MME support */ +#include + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audio_c.h" +#include "SDL_mmeaudio.h" + +static BOOL inUse[NUM_BUFFERS]; + +/* Audio driver functions */ +static int MME_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void MME_WaitAudio(_THIS); +static Uint8 *MME_GetAudioBuf(_THIS); +static void MME_PlayAudio(_THIS); +static void MME_WaitDone(_THIS); +static void MME_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ +static int Audio_Available(void) +{ + return(1); +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + if ( device ) { + if ( device->hidden ) { + SDL_free(device->hidden); + device->hidden = NULL; + } + SDL_free(device); + device = NULL; + } +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + +/* Initialize all variables that we clean on shutdown */ + this = SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + /* Set the function pointers */ + this->OpenAudio = MME_OpenAudio; + this->WaitAudio = MME_WaitAudio; + this->PlayAudio = MME_PlayAudio; + this->GetAudioBuf = MME_GetAudioBuf; + this->WaitDone = MME_WaitDone; + this->CloseAudio = MME_CloseAudio; + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap MMEAUDIO_bootstrap = { + "waveout", "Tru64 MME WaveOut", + Audio_Available, Audio_CreateDevice +}; + +static void SetMMerror(char *function, MMRESULT code) +{ + int len; + char errbuf[MAXERRORLENGTH]; + + SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: ", function); + len = SDL_strlen(errbuf); + waveOutGetErrorText(code, errbuf+len, MAXERRORLENGTH-len); + SDL_SetError("%s",errbuf); +} + +static void CALLBACK MME_CALLBACK(HWAVEOUT hwo, + UINT uMsg, + DWORD dwInstance, + LPARAM dwParam1, + LPARAM dwParam2) +{ + WAVEHDR *wp = (WAVEHDR *) dwParam1; + + if ( uMsg == WOM_DONE ) + inUse[wp->dwUser] = FALSE; +} + +static int MME_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + MMRESULT result; + int i; + + mixbuf = NULL; + + /* Set basic WAVE format parameters */ + shm = mmeAllocMem(sizeof(*shm)); + if ( shm == NULL ) { + SDL_SetError("Out of memory: shm"); + return(-1); + } + shm->sound = 0; + shm->wFmt.wf.wFormatTag = WAVE_FORMAT_PCM; + + /* Determine the audio parameters from the AudioSpec */ + switch ( spec->format & 0xFF ) { + case 8: + /* Unsigned 8 bit audio data */ + spec->format = AUDIO_U8; + shm->wFmt.wBitsPerSample = 8; + break; + case 16: + /* Signed 16 bit audio data */ + spec->format = AUDIO_S16; + shm->wFmt.wBitsPerSample = 16; + break; + default: + SDL_SetError("Unsupported audio format"); + return(-1); + } + + shm->wFmt.wf.nChannels = spec->channels; + shm->wFmt.wf.nSamplesPerSec = spec->freq; + shm->wFmt.wf.nBlockAlign = + shm->wFmt.wf.nChannels * shm->wFmt.wBitsPerSample / 8; + shm->wFmt.wf.nAvgBytesPerSec = + shm->wFmt.wf.nSamplesPerSec * shm->wFmt.wf.nBlockAlign; + + /* Check the buffer size -- minimum of 1/4 second (word aligned) */ + if ( spec->samples < (spec->freq/4) ) + spec->samples = ((spec->freq/4)+3)&~3; + + /* Update the fragment size as size in bytes */ + SDL_CalculateAudioSpec(spec); + + /* Open the audio device */ + result = waveOutOpen(&(shm->sound), + WAVE_MAPPER, + &(shm->wFmt.wf), + MME_CALLBACK, + NULL, + (CALLBACK_FUNCTION|WAVE_OPEN_SHAREABLE)); + if ( result != MMSYSERR_NOERROR ) { + SetMMerror("waveOutOpen()", result); + return(-1); + } + + /* Create the sound buffers */ + mixbuf = (Uint8 *)mmeAllocBuffer(NUM_BUFFERS * (spec->size)); + if ( mixbuf == NULL ) { + SDL_SetError("Out of memory: mixbuf"); + return(-1); + } + + for (i = 0; i < NUM_BUFFERS; i++) { + shm->wHdr[i].lpData = &mixbuf[i * (spec->size)]; + shm->wHdr[i].dwBufferLength = spec->size; + shm->wHdr[i].dwFlags = 0; + shm->wHdr[i].dwUser = i; + shm->wHdr[i].dwLoops = 0; /* loop control counter */ + shm->wHdr[i].lpNext = NULL; /* reserved for driver */ + shm->wHdr[i].reserved = 0; + inUse[i] = FALSE; + } + next_buffer = 0; + return 0; +} + +static void MME_WaitAudio(_THIS) +{ + while ( inUse[next_buffer] ) { + mmeWaitForCallbacks(); + mmeProcessCallbacks(); + } +} + +static Uint8 *MME_GetAudioBuf(_THIS) +{ + Uint8 *retval; + + inUse[next_buffer] = TRUE; + retval = (Uint8 *)(shm->wHdr[next_buffer].lpData); + return retval; +} + +static void MME_PlayAudio(_THIS) +{ + /* Queue it up */ + waveOutWrite(shm->sound, &(shm->wHdr[next_buffer]), sizeof(WAVEHDR)); + next_buffer = (next_buffer+1)%NUM_BUFFERS; +} + +static void MME_WaitDone(_THIS) +{ + MMRESULT result; + int i; + + if ( shm->sound ) { + for (i = 0; i < NUM_BUFFERS; i++) + while ( inUse[i] ) { + mmeWaitForCallbacks(); + mmeProcessCallbacks(); + } + result = waveOutReset(shm->sound); + if ( result != MMSYSERR_NOERROR ) + SetMMerror("waveOutReset()", result); + mmeProcessCallbacks(); + } +} + +static void MME_CloseAudio(_THIS) +{ + MMRESULT result; + + if ( mixbuf ) { + result = mmeFreeBuffer(mixbuf); + if (result != MMSYSERR_NOERROR ) + SetMMerror("mmeFreeBuffer", result); + mixbuf = NULL; + } + + if ( shm ) { + if ( shm->sound ) { + result = waveOutClose(shm->sound); + if (result != MMSYSERR_NOERROR ) + SetMMerror("waveOutClose()", result); + mmeProcessCallbacks(); + } + result = mmeFreeMem(shm); + if (result != MMSYSERR_NOERROR ) + SetMMerror("mmeFreeMem()", result); + shm = NULL; + } +} + diff --git a/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h b/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h new file mode 100644 index 0000000000..6bfaed32e0 --- /dev/null +++ b/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h @@ -0,0 +1,51 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + + Sam Lantinga + slouken@libsdl.org +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#ifndef _SDL_lowaudio_h +#define _SDL_lowaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this +#define NUM_BUFFERS 2 + +struct SharedMem { + HWAVEOUT sound; + WAVEHDR wHdr[NUM_BUFFERS]; + PCMWAVEFORMAT wFmt; +}; + +struct SDL_PrivateAudioData { + Uint8 *mixbuf; /* The raw allocated mixing buffer */ + struct SharedMem *shm; + int next_buffer; +}; + +#define shm (this->hidden->shm) +#define mixbuf (this->hidden->mixbuf) +#define next_buffer (this->hidden->next_buffer) +/* Old variable names */ +#endif /* _SDL_lowaudio_h */ diff --git a/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c b/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c new file mode 100644 index 0000000000..a561e62984 --- /dev/null +++ b/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c @@ -0,0 +1,423 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org + + This driver was written by: + Erik Inge Bolsø + knan@mo.himolde.no +*/ +#include "SDL_config.h" + +/* Allow access to a raw mixing buffer */ + +#include +#include + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_nasaudio.h" + +#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC +#include "SDL_loadso.h" +#endif + +/* The tag name used by artsc audio */ +#define NAS_DRIVER_NAME "nas" + +static struct SDL_PrivateAudioData *this2 = NULL; + +static void (*NAS_AuCloseServer) (AuServer *); +static void (*NAS_AuNextEvent) (AuServer *, AuBool, AuEvent *); +static AuBool(*NAS_AuDispatchEvent) (AuServer *, AuEvent *); +static AuFlowID(*NAS_AuCreateFlow) (AuServer *, AuStatus *); +static void (*NAS_AuStartFlow) (AuServer *, AuFlowID, AuStatus *); +static void (*NAS_AuSetElements) + (AuServer *, AuFlowID, AuBool, int, AuElement *, AuStatus *); +static void (*NAS_AuWriteElement) + (AuServer *, AuFlowID, int, AuUint32, AuPointer, AuBool, AuStatus *); +static AuServer *(*NAS_AuOpenServer) + (_AuConst char *, int, _AuConst char *, int, _AuConst char *, char **); +static AuEventHandlerRec *(*NAS_AuRegisterEventHandler) + (AuServer *, AuMask, int, AuID, AuEventHandlerCallback, AuPointer); + + +#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC + +static const char *nas_library = SDL_AUDIO_DRIVER_NAS_DYNAMIC; +static void *nas_handle = NULL; + +static int +load_nas_sym(const char *fn, void **addr) +{ + *addr = SDL_LoadFunction(nas_handle, fn); + if (*addr == NULL) { + return 0; + } + return 1; +} + +/* cast funcs to char* first, to please GCC's strict aliasing rules. */ +#define SDL_NAS_SYM(x) \ + if (!load_nas_sym(#x, (void **) (char *) &NAS_##x)) return -1 +#else +#define SDL_NAS_SYM(x) NAS_##x = x +#endif + +static int +load_nas_syms(void) +{ + SDL_NAS_SYM(AuCloseServer); + SDL_NAS_SYM(AuNextEvent); + SDL_NAS_SYM(AuDispatchEvent); + SDL_NAS_SYM(AuCreateFlow); + SDL_NAS_SYM(AuStartFlow); + SDL_NAS_SYM(AuSetElements); + SDL_NAS_SYM(AuWriteElement); + SDL_NAS_SYM(AuOpenServer); + SDL_NAS_SYM(AuRegisterEventHandler); + return 0; +} + +#undef SDL_NAS_SYM + +#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC + +static void +UnloadNASLibrary(void) +{ + if (nas_handle != NULL) { + SDL_UnloadObject(nas_handle); + nas_handle = NULL; + } +} + +static int +LoadNASLibrary(void) +{ + int retval = 0; + if (nas_handle == NULL) { + nas_handle = SDL_LoadObject(nas_library); + if (nas_handle == NULL) { + /* Copy error string so we can use it in a new SDL_SetError(). */ + char *origerr = SDL_GetError(); + size_t len = SDL_strlen(origerr) + 1; + char *err = (char *) alloca(len); + SDL_strlcpy(err, origerr, len); + retval = -1; + SDL_SetError("NAS: SDL_LoadObject('%s') failed: %s\n", + nas_library, err); + } else { + retval = load_nas_syms(); + if (retval < 0) { + UnloadNASLibrary(); + } + } + } + return retval; +} + +#else + +static void +UnloadNASLibrary(void) +{ +} + +static int +LoadNASLibrary(void) +{ + load_nas_syms(); + return 0; +} + +#endif /* SDL_AUDIO_DRIVER_NAS_DYNAMIC */ + + +/* Audio driver functions */ +static int NAS_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void NAS_WaitAudio(_THIS); +static void NAS_PlayAudio(_THIS); +static Uint8 *NAS_GetAudioBuf(_THIS); +static void NAS_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ + +static int Audio_Available(void) +{ + if (LoadNASLibrary() == 0) { + AuServer *aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL); + if (!aud) { + UnloadNASLibrary(); + return 0; + } + NAS_AuCloseServer(aud); + UnloadNASLibrary(); + return 1; + } + return 0; +} + +static void Audio_DeleteDevice(SDL_AudioDevice *device) +{ + UnloadNASLibrary(); + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *Audio_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + if (LoadNASLibrary() < 0) { + return NULL; + } + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return NULL; + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = NAS_OpenAudio; + this->WaitAudio = NAS_WaitAudio; + this->PlayAudio = NAS_PlayAudio; + this->GetAudioBuf = NAS_GetAudioBuf; + this->CloseAudio = NAS_CloseAudio; + + this->free = Audio_DeleteDevice; + + return this; +} + +AudioBootStrap NAS_bootstrap = { + NAS_DRIVER_NAME, "Network Audio System", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void NAS_WaitAudio(_THIS) +{ + while ( this->hidden->buf_free < this->hidden->mixlen ) { + AuEvent ev; + NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev); + NAS_AuDispatchEvent(this->hidden->aud, &ev); + } +} + +static void NAS_PlayAudio(_THIS) +{ + while (this->hidden->mixlen > this->hidden->buf_free) { /* We think the buffer is full? Yikes! Ask the server for events, + in the hope that some of them is LowWater events telling us more + of the buffer is free now than what we think. */ + AuEvent ev; + NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev); + NAS_AuDispatchEvent(this->hidden->aud, &ev); + } + this->hidden->buf_free -= this->hidden->mixlen; + + /* Write the audio data */ + NAS_AuWriteElement(this->hidden->aud, this->hidden->flow, 0, this->hidden->mixlen, this->hidden->mixbuf, AuFalse, NULL); + + this->hidden->written += this->hidden->mixlen; + +#ifdef DEBUG_AUDIO + fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen); +#endif +} + +static Uint8 *NAS_GetAudioBuf(_THIS) +{ + return(this->hidden->mixbuf); +} + +static void NAS_CloseAudio(_THIS) +{ + if ( this->hidden->mixbuf != NULL ) { + SDL_FreeAudioMem(this->hidden->mixbuf); + this->hidden->mixbuf = NULL; + } + if ( this->hidden->aud ) { + NAS_AuCloseServer(this->hidden->aud); + this->hidden->aud = 0; + } +} + +static unsigned char sdlformat_to_auformat(unsigned int fmt) +{ + switch (fmt) + { + case AUDIO_U8: + return AuFormatLinearUnsigned8; + case AUDIO_S8: + return AuFormatLinearSigned8; + case AUDIO_U16LSB: + return AuFormatLinearUnsigned16LSB; + case AUDIO_U16MSB: + return AuFormatLinearUnsigned16MSB; + case AUDIO_S16LSB: + return AuFormatLinearSigned16LSB; + case AUDIO_S16MSB: + return AuFormatLinearSigned16MSB; + } + return AuNone; +} + +static AuBool +event_handler(AuServer* aud, AuEvent* ev, AuEventHandlerRec* hnd) +{ + switch (ev->type) { + case AuEventTypeElementNotify: { + AuElementNotifyEvent* event = (AuElementNotifyEvent *)ev; + + switch (event->kind) { + case AuElementNotifyKindLowWater: + if (this2->buf_free >= 0) { + this2->really += event->num_bytes; + gettimeofday(&this2->last_tv, 0); + this2->buf_free += event->num_bytes; + } else { + this2->buf_free = event->num_bytes; + } + break; + case AuElementNotifyKindState: + switch (event->cur_state) { + case AuStatePause: + if (event->reason != AuReasonUser) { + if (this2->buf_free >= 0) { + this2->really += event->num_bytes; + gettimeofday(&this2->last_tv, 0); + this2->buf_free += event->num_bytes; + } else { + this2->buf_free = event->num_bytes; + } + } + break; + } + } + } + } + return AuTrue; +} + +static AuDeviceID +find_device(_THIS, int nch) +{ + /* These "Au" things are all macros, not functions... */ + int i; + for (i = 0; i < AuServerNumDevices(this->hidden->aud); i++) { + if ((AuDeviceKind(AuServerDevice(this->hidden->aud, i)) == + AuComponentKindPhysicalOutput) && + AuDeviceNumTracks(AuServerDevice(this->hidden->aud, i)) == nch) { + return AuDeviceIdentifier(AuServerDevice(this->hidden->aud, i)); + } + } + return AuNone; +} + +static int NAS_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + AuElement elms[3]; + int buffer_size; + Uint16 test_format, format; + + this->hidden->mixbuf = NULL; + + /* Try for a closest match on audio format */ + format = 0; + for ( test_format = SDL_FirstAudioFormat(spec->format); + ! format && test_format; ) { + format = sdlformat_to_auformat(test_format); + + if (format == AuNone) { + test_format = SDL_NextAudioFormat(); + } + } + if ( format == 0 ) { + SDL_SetError("Couldn't find any hardware audio formats"); + return(-1); + } + spec->format = test_format; + + this->hidden->aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL); + if (this->hidden->aud == 0) + { + SDL_SetError("Couldn't open connection to NAS server"); + return (-1); + } + + this->hidden->dev = find_device(this, spec->channels); + if ((this->hidden->dev == AuNone) || (!(this->hidden->flow = NAS_AuCreateFlow(this->hidden->aud, NULL)))) { + NAS_AuCloseServer(this->hidden->aud); + this->hidden->aud = 0; + SDL_SetError("Couldn't find a fitting playback device on NAS server"); + return (-1); + } + + buffer_size = spec->freq; + if (buffer_size < 4096) + buffer_size = 4096; + + if (buffer_size > 32768) + buffer_size = 32768; /* So that the buffer won't get unmanageably big. */ + + /* Calculate the final parameters for this audio specification */ + SDL_CalculateAudioSpec(spec); + + this2 = this->hidden; + + /* These "Au" things without a NAS_ prefix are macros, not functions... */ + AuMakeElementImportClient(elms, spec->freq, format, spec->channels, AuTrue, + buffer_size, buffer_size / 4, 0, NULL); + AuMakeElementExportDevice(elms+1, 0, this->hidden->dev, spec->freq, + AuUnlimitedSamples, 0, NULL); + NAS_AuSetElements(this->hidden->aud, this->hidden->flow, AuTrue, 2, elms, NULL); + NAS_AuRegisterEventHandler(this->hidden->aud, AuEventHandlerIDMask, 0, this->hidden->flow, + event_handler, (AuPointer) NULL); + + NAS_AuStartFlow(this->hidden->aud, this->hidden->flow, NULL); + + /* Allocate mixing buffer */ + this->hidden->mixlen = spec->size; + this->hidden->mixbuf = (Uint8 *)SDL_AllocAudioMem(this->hidden->mixlen); + if ( this->hidden->mixbuf == NULL ) { + return(-1); + } + SDL_memset(this->hidden->mixbuf, spec->silence, spec->size); + + /* Get the parent process id (we're the parent of the audio thread) */ + this->hidden->parent = getpid(); + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h b/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h new file mode 100644 index 0000000000..1c09630880 --- /dev/null +++ b/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h @@ -0,0 +1,62 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 Sam Lantinga + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Lesser General Public + License as published by the Free Software Foundation; either + version 2.1 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Lesser General Public License for more details. + + You should have received a copy of the GNU Lesser General Public + License along with this library; if not, write to the Free Software + Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + Sam Lantinga + slouken@libsdl.org + + This driver was written by: + Erik Inge Bolsø + knan@mo.himolde.no +*/ +#include "SDL_config.h" + +#ifndef _SDL_nasaudio_h +#define _SDL_nasaudio_h + +#ifdef __sgi +#include +#else +#include