--HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%403404
1236 lines
36 KiB
C
1236 lines
36 KiB
C
/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2009 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
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License along with this library; if not, write to the Free Software
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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/* Allow access to a raw mixing buffer */
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#include "SDL.h"
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#include "SDL_audio_c.h"
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#include "SDL_audiomem.h"
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#include "SDL_sysaudio.h"
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#define _THIS SDL_AudioDevice *this
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static SDL_AudioDriver current_audio;
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static SDL_AudioDevice *open_devices[16];
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/* !!! FIXME: These are wordy and unlocalized... */
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#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
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#define DEFAULT_INPUT_DEVNAME "System audio capture device"
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/*
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* Not all of these will be compiled and linked in, but it's convenient
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* to have a complete list here and saves yet-another block of #ifdefs...
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* Please see bootstrap[], below, for the actual #ifdef mess.
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*/
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extern AudioBootStrap BSD_AUDIO_bootstrap;
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extern AudioBootStrap DSP_bootstrap;
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extern AudioBootStrap DMA_bootstrap;
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extern AudioBootStrap ALSA_bootstrap;
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extern AudioBootStrap PULSEAUDIO_bootstrap;
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extern AudioBootStrap QNXNTOAUDIO_bootstrap;
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extern AudioBootStrap SUNAUDIO_bootstrap;
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extern AudioBootStrap DMEDIA_bootstrap;
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extern AudioBootStrap ARTS_bootstrap;
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extern AudioBootStrap ESD_bootstrap;
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extern AudioBootStrap NAS_bootstrap;
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extern AudioBootStrap DSOUND_bootstrap;
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extern AudioBootStrap WINWAVEOUT_bootstrap;
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extern AudioBootStrap PAUDIO_bootstrap;
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extern AudioBootStrap BEOSAUDIO_bootstrap;
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extern AudioBootStrap COREAUDIO_bootstrap;
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extern AudioBootStrap COREAUDIOIPHONE_bootstrap;
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extern AudioBootStrap SNDMGR_bootstrap;
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extern AudioBootStrap MINTAUDIO_GSXB_bootstrap;
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extern AudioBootStrap MINTAUDIO_MCSN_bootstrap;
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extern AudioBootStrap MINTAUDIO_STFA_bootstrap;
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extern AudioBootStrap MINTAUDIO_XBIOS_bootstrap;
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extern AudioBootStrap MINTAUDIO_DMA8_bootstrap;
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extern AudioBootStrap DISKAUD_bootstrap;
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extern AudioBootStrap DUMMYAUD_bootstrap;
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extern AudioBootStrap DCAUD_bootstrap;
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extern AudioBootStrap MMEAUDIO_bootstrap;
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extern AudioBootStrap DART_bootstrap;
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extern AudioBootStrap NDSAUD_bootstrap;
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/* Available audio drivers */
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static AudioBootStrap *bootstrap[] = {
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#if SDL_AUDIO_DRIVER_BSD
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&BSD_AUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_PULSEAUDIO
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&PULSEAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ALSA
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&ALSA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_OSS
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&DSP_bootstrap,
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&DMA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_QNXNTO
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&QNXNTOAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_SUNAUDIO
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&SUNAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DMEDIA
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&DMEDIA_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ARTS
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&ARTS_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ESD
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&ESD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_NAS
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&NAS_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DSOUND
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&DSOUND_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_WINWAVEOUT
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&WINWAVEOUT_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_PAUDIO
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&PAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_BEOSAUDIO
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&BEOSAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_COREAUDIO
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&COREAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_COREAUDIOIPHONE
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&COREAUDIOIPHONE_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_SNDMGR
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&SNDMGR_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_MINT
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&MINTAUDIO_GSXB_bootstrap,
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&MINTAUDIO_MCSN_bootstrap,
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&MINTAUDIO_STFA_bootstrap,
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&MINTAUDIO_XBIOS_bootstrap,
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&MINTAUDIO_DMA8_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DISK
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&DISKAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DUMMY
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&DUMMYAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DC
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&DCAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_MMEAUDIO
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&MMEAUDIO_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DART
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&DART_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_NDS
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&NDSAUD_bootstrap,
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#endif
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NULL
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};
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static SDL_AudioDevice *
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get_audio_device(SDL_AudioDeviceID id)
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{
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id--;
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if ((id >= SDL_arraysize(open_devices)) || (open_devices[id] == NULL)) {
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SDL_SetError("Invalid audio device ID");
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return NULL;
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}
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return open_devices[id];
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}
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/* stubs for audio drivers that don't need a specific entry point... */
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static int
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SDL_AudioDetectDevices_Default(int iscapture)
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{
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return -1;
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}
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static void
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SDL_AudioThreadInit_Default(_THIS)
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{ /* no-op. */
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}
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static void
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SDL_AudioWaitDevice_Default(_THIS)
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{ /* no-op. */
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}
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static void
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SDL_AudioPlayDevice_Default(_THIS)
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{ /* no-op. */
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}
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static Uint8 *
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SDL_AudioGetDeviceBuf_Default(_THIS)
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{
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return NULL;
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}
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static void
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SDL_AudioWaitDone_Default(_THIS)
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{ /* no-op. */
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}
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static void
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SDL_AudioCloseDevice_Default(_THIS)
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{ /* no-op. */
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}
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static void
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SDL_AudioDeinitialize_Default(void)
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{ /* no-op. */
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}
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static int
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SDL_AudioOpenDevice_Default(_THIS, const char *devname, int iscapture)
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{
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return 0;
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}
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static const char *
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SDL_AudioGetDeviceName_Default(int index, int iscapture)
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{
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SDL_SetError("No such device");
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return NULL;
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}
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static void
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SDL_AudioLockDevice_Default(SDL_AudioDevice * device)
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{
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if (device->thread && (SDL_ThreadID() == device->threadid)) {
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return;
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}
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SDL_mutexP(device->mixer_lock);
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}
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static void
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SDL_AudioUnlockDevice_Default(SDL_AudioDevice * device)
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{
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if (device->thread && (SDL_ThreadID() == device->threadid)) {
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return;
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}
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SDL_mutexV(device->mixer_lock);
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}
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static void
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finalize_audio_entry_points(void)
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{
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/*
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* Fill in stub functions for unused driver entry points. This lets us
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* blindly call them without having to check for validity first.
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*/
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#define FILL_STUB(x) \
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if (current_audio.impl.x == NULL) { \
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current_audio.impl.x = SDL_Audio##x##_Default; \
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}
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FILL_STUB(DetectDevices);
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FILL_STUB(GetDeviceName);
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FILL_STUB(OpenDevice);
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FILL_STUB(ThreadInit);
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FILL_STUB(WaitDevice);
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FILL_STUB(PlayDevice);
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FILL_STUB(GetDeviceBuf);
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FILL_STUB(WaitDone);
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FILL_STUB(CloseDevice);
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FILL_STUB(LockDevice);
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FILL_STUB(UnlockDevice);
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FILL_STUB(Deinitialize);
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#undef FILL_STUB
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}
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/* Streaming functions (for when the input and output buffer sizes are different) */
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/* Write [length] bytes from buf into the streamer */
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void
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SDL_StreamWrite(SDL_AudioStreamer * stream, Uint8 * buf, int length)
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{
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int i;
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for (i = 0; i < length; ++i) {
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stream->buffer[stream->write_pos] = buf[i];
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++stream->write_pos;
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}
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}
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/* Read [length] bytes out of the streamer into buf */
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void
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SDL_StreamRead(SDL_AudioStreamer * stream, Uint8 * buf, int length)
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{
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int i;
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for (i = 0; i < length; ++i) {
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buf[i] = stream->buffer[stream->read_pos];
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++stream->read_pos;
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}
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}
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int
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SDL_StreamLength(SDL_AudioStreamer * stream)
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{
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return (stream->write_pos - stream->read_pos) % stream->max_len;
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}
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/* Initialize the stream by allocating the buffer and setting the read/write heads to the beginning */
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int
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SDL_StreamInit(SDL_AudioStreamer * stream, int max_len, Uint8 silence)
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{
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/* First try to allocate the buffer */
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stream->buffer = (Uint8 *) SDL_malloc(max_len);
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if (stream->buffer == NULL) {
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return -1;
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}
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stream->max_len = max_len;
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stream->read_pos = 0;
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stream->write_pos = 0;
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/* Zero out the buffer */
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SDL_memset(stream->buffer, silence, max_len);
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return 0;
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}
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/* Deinitialize the stream simply by freeing the buffer */
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void
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SDL_StreamDeinit(SDL_AudioStreamer * stream)
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{
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if (stream->buffer != NULL) {
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SDL_free(stream->buffer);
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}
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}
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/* The general mixing thread function */
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int SDLCALL
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SDL_RunAudio(void *devicep)
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{
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SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
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Uint8 *stream;
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int stream_len;
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void *udata;
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void (SDLCALL * fill) (void *userdata, Uint8 * stream, int len);
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int silence;
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int stream_max_len;
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/* For streaming when the buffer sizes don't match up */
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Uint8 *istream;
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int istream_len;
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/* Perform any thread setup */
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device->threadid = SDL_ThreadID();
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current_audio.impl.ThreadInit(device);
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/* Set up the mixing function */
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fill = device->spec.callback;
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udata = device->spec.userdata;
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/* By default do not stream */
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device->use_streamer = 0;
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if (device->convert.needed) {
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if (device->convert.src_format == AUDIO_U8) {
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silence = 0x80;
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} else {
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silence = 0;
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}
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/* If the result of the conversion alters the length, i.e. resampling is being used, use the streamer */
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if (device->convert.len_mult != 1 || device->convert.len_div != 1) {
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/* The streamer's maximum length should be twice whichever is larger: spec.size or len_cvt */
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stream_max_len = 2 * device->spec.size;
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if (device->convert.len_mult > device->convert.len_div) {
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stream_max_len *= device->convert.len_mult;
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stream_max_len /= device->convert.len_div;
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}
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if (SDL_StreamInit(&device->streamer, stream_max_len, silence) <
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0)
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return -1;
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device->use_streamer = 1;
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/* istream_len should be the length of what we grab from the callback and feed to conversion,
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so that we get close to spec_size. I.e. we want device.spec_size = istream_len * u / d
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*/
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istream_len =
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device->spec.size * device->convert.len_div /
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device->convert.len_mult;
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}
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/* stream_len = device->convert.len; */
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stream_len = device->spec.size;
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} else {
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silence = device->spec.silence;
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stream_len = device->spec.size;
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}
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/* Determine if the streamer is necessary here */
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if (device->use_streamer == 1) {
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/* This code is almost the same as the old code. The difference is, instead of reding
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directly from the callback into "stream", then converting and sending the audio off,
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we go: callback -> "istream" -> (conversion) -> streamer -> stream -> device.
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However, reading and writing with streamer are done separately:
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- We only call the callback and write to the streamer when the streamer does not
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contain enough samples to output to the device.
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- We only read from the streamer and tell the device to play when the streamer
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does have enough samples to output.
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This allows us to perform resampling in the conversion step, where the output of the
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resampling process can be any number. We will have to see what a good size for the
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stream's maximum length is, but I suspect 2*max(len_cvt, stream_len) is a good figure.
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*/
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while (device->enabled) {
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/* Only read in audio if the streamer doesn't have enough already (if it does not have enough samples to output) */
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if (SDL_StreamLength(&device->streamer) < stream_len) {
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/* Set up istream */
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if (device->convert.needed) {
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if (device->convert.buf) {
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istream = device->convert.buf;
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} else {
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continue;
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}
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} else {
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istream = current_audio.impl.GetDeviceBuf(device);
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if (istream == NULL) {
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istream = device->fake_stream;
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}
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}
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/* Read from the callback into the _input_ stream */
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if (!device->paused) {
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SDL_mutexP(device->mixer_lock);
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(*fill) (udata, istream, istream_len);
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SDL_mutexV(device->mixer_lock);
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}
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/* Convert the audio if necessary and write to the streamer */
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if (device->convert.needed) {
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SDL_ConvertAudio(&device->convert);
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if (istream == NULL) {
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istream = device->fake_stream;
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}
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/*SDL_memcpy(istream, device->convert.buf, device->convert.len_cvt); */
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SDL_StreamWrite(&device->streamer, device->convert.buf,
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device->convert.len_cvt);
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} else {
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SDL_StreamWrite(&device->streamer, istream, istream_len);
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}
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}
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/* Only output audio if the streamer has enough to output */
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if (SDL_StreamLength(&device->streamer) >= stream_len) {
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/* Set up the output stream */
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if (device->convert.needed) {
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if (device->convert.buf) {
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stream = device->convert.buf;
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} else {
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continue;
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}
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} else {
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stream = current_audio.impl.GetDeviceBuf(device);
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if (stream == NULL) {
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stream = device->fake_stream;
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}
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}
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/* Now read from the streamer */
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SDL_StreamRead(&device->streamer, stream, stream_len);
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/* Ready current buffer for play and change current buffer */
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if (stream != device->fake_stream) {
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current_audio.impl.PlayDevice(device);
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}
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/* Wait for an audio buffer to become available */
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if (stream == device->fake_stream) {
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SDL_Delay((device->spec.samples * 1000) /
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device->spec.freq);
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} else {
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current_audio.impl.WaitDevice(device);
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}
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}
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}
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} else {
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/* Otherwise, do not use the streamer. This is the old code. */
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/* Loop, filling the audio buffers */
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while (device->enabled) {
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/* Fill the current buffer with sound */
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if (device->convert.needed) {
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if (device->convert.buf) {
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stream = device->convert.buf;
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} else {
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continue;
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}
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} else {
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stream = current_audio.impl.GetDeviceBuf(device);
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if (stream == NULL) {
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stream = device->fake_stream;
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}
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}
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if (!device->paused) {
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SDL_mutexP(device->mixer_lock);
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(*fill) (udata, stream, stream_len);
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SDL_mutexV(device->mixer_lock);
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}
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/* Convert the audio if necessary */
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if (device->convert.needed) {
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SDL_ConvertAudio(&device->convert);
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stream = current_audio.impl.GetDeviceBuf(device);
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if (stream == NULL) {
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stream = device->fake_stream;
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}
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SDL_memcpy(stream, device->convert.buf,
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device->convert.len_cvt);
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}
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/* Ready current buffer for play and change current buffer */
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if (stream != device->fake_stream) {
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current_audio.impl.PlayDevice(device);
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}
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|
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/* Wait for an audio buffer to become available */
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if (stream == device->fake_stream) {
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SDL_Delay((device->spec.samples * 1000) / device->spec.freq);
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} else {
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current_audio.impl.WaitDevice(device);
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}
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}
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}
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|
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/* Wait for the audio to drain.. */
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current_audio.impl.WaitDone(device);
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|
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/* If necessary, deinit the streamer */
|
|
if (device->use_streamer == 1)
|
|
SDL_StreamDeinit(&device->streamer);
|
|
|
|
return (0);
|
|
}
|
|
|
|
|
|
static SDL_AudioFormat
|
|
SDL_ParseAudioFormat(const char *string)
|
|
{
|
|
#define CHECK_FMT_STRING(x) if (SDL_strcmp(string, #x) == 0) return AUDIO_##x
|
|
CHECK_FMT_STRING(U8);
|
|
CHECK_FMT_STRING(S8);
|
|
CHECK_FMT_STRING(U16LSB);
|
|
CHECK_FMT_STRING(S16LSB);
|
|
CHECK_FMT_STRING(U16MSB);
|
|
CHECK_FMT_STRING(S16MSB);
|
|
CHECK_FMT_STRING(U16SYS);
|
|
CHECK_FMT_STRING(S16SYS);
|
|
CHECK_FMT_STRING(U16);
|
|
CHECK_FMT_STRING(S16);
|
|
CHECK_FMT_STRING(S32LSB);
|
|
CHECK_FMT_STRING(S32MSB);
|
|
CHECK_FMT_STRING(S32SYS);
|
|
CHECK_FMT_STRING(S32);
|
|
CHECK_FMT_STRING(F32LSB);
|
|
CHECK_FMT_STRING(F32MSB);
|
|
CHECK_FMT_STRING(F32SYS);
|
|
CHECK_FMT_STRING(F32);
|
|
#undef CHECK_FMT_STRING
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
SDL_GetNumAudioDrivers(void)
|
|
{
|
|
return (SDL_arraysize(bootstrap) - 1);
|
|
}
|
|
|
|
const char *
|
|
SDL_GetAudioDriver(int index)
|
|
{
|
|
if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
|
|
return (bootstrap[index]->name);
|
|
}
|
|
return (NULL);
|
|
}
|
|
|
|
int
|
|
SDL_AudioInit(const char *driver_name)
|
|
{
|
|
int i = 0;
|
|
int initialized = 0;
|
|
int tried_to_init = 0;
|
|
int rc = 0;
|
|
int best_choice = -1;
|
|
|
|
if (SDL_WasInit(SDL_INIT_AUDIO)) {
|
|
SDL_AudioQuit(); /* shutdown driver if already running. */
|
|
}
|
|
|
|
SDL_memset(¤t_audio, '\0', sizeof(current_audio));
|
|
SDL_memset(open_devices, '\0', sizeof(open_devices));
|
|
|
|
/* Select the proper audio driver */
|
|
if (driver_name == NULL) {
|
|
driver_name = SDL_getenv("SDL_AUDIODRIVER");
|
|
}
|
|
|
|
for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
|
|
/* make sure we should even try this driver before doing so... */
|
|
const AudioBootStrap *backend = bootstrap[i];
|
|
if (((driver_name) && (SDL_strcasecmp(backend->name, driver_name))) ||
|
|
((!driver_name) && (backend->demand_only))) {
|
|
continue;
|
|
}
|
|
|
|
tried_to_init = 1;
|
|
SDL_memset(¤t_audio, 0, sizeof(current_audio));
|
|
current_audio.name = backend->name;
|
|
current_audio.desc = backend->desc;
|
|
rc = backend->init(¤t_audio.impl);
|
|
if (rc == 2) { /* init'd, and devices available. Take it! */
|
|
initialized = 1;
|
|
best_choice = i;
|
|
} else if (rc == 1) { /* init'd, but can't see any devices. */
|
|
current_audio.impl.Deinitialize();
|
|
if (best_choice == -1) {
|
|
best_choice = i;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* No definite choice. Pick one that works but can't promise a device. */
|
|
if ((!initialized) && (best_choice != -1)) {
|
|
const AudioBootStrap *backend = bootstrap[best_choice];
|
|
SDL_memset(¤t_audio, 0, sizeof(current_audio));
|
|
current_audio.name = backend->name;
|
|
current_audio.desc = backend->desc;
|
|
initialized = (backend->init(¤t_audio.impl) > 0);
|
|
}
|
|
|
|
if (!initialized) {
|
|
/* specific drivers will set the error message if they fail... */
|
|
if (!tried_to_init) {
|
|
if (driver_name) {
|
|
SDL_SetError("%s not available", driver_name);
|
|
} else {
|
|
SDL_SetError("No available audio device");
|
|
}
|
|
}
|
|
|
|
SDL_memset(¤t_audio, 0, sizeof(current_audio));
|
|
return (-1); /* No driver was available, so fail. */
|
|
}
|
|
|
|
finalize_audio_entry_points();
|
|
|
|
return (0);
|
|
}
|
|
|
|
/*
|
|
* Get the current audio driver name
|
|
*/
|
|
const char *
|
|
SDL_GetCurrentAudioDriver()
|
|
{
|
|
return current_audio.name;
|
|
}
|
|
|
|
|
|
int
|
|
SDL_GetNumAudioDevices(int iscapture)
|
|
{
|
|
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
|
|
return -1;
|
|
}
|
|
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
|
|
return 0;
|
|
}
|
|
|
|
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
|
|
return 1;
|
|
}
|
|
|
|
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
|
|
return 1;
|
|
}
|
|
|
|
return current_audio.impl.DetectDevices(iscapture);
|
|
}
|
|
|
|
|
|
const char *
|
|
SDL_GetAudioDeviceName(int index, int iscapture)
|
|
{
|
|
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
|
|
SDL_SetError("Audio subsystem is not initialized");
|
|
return NULL;
|
|
}
|
|
|
|
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
|
|
SDL_SetError("No capture support");
|
|
return NULL;
|
|
}
|
|
|
|
if (index < 0) {
|
|
SDL_SetError("No such device");
|
|
return NULL;
|
|
}
|
|
|
|
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
|
|
return DEFAULT_INPUT_DEVNAME;
|
|
}
|
|
|
|
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
|
|
return DEFAULT_OUTPUT_DEVNAME;
|
|
}
|
|
|
|
return current_audio.impl.GetDeviceName(index, iscapture);
|
|
}
|
|
|
|
|
|
static void
|
|
close_audio_device(SDL_AudioDevice * device)
|
|
{
|
|
device->enabled = 0;
|
|
if (device->thread != NULL) {
|
|
SDL_WaitThread(device->thread, NULL);
|
|
}
|
|
if (device->mixer_lock != NULL) {
|
|
SDL_DestroyMutex(device->mixer_lock);
|
|
}
|
|
if (device->fake_stream != NULL) {
|
|
SDL_FreeAudioMem(device->fake_stream);
|
|
}
|
|
if (device->convert.needed) {
|
|
SDL_FreeAudioMem(device->convert.buf);
|
|
}
|
|
if (device->opened) {
|
|
current_audio.impl.CloseDevice(device);
|
|
device->opened = 0;
|
|
}
|
|
SDL_FreeAudioMem(device);
|
|
}
|
|
|
|
|
|
/*
|
|
* Sanity check desired AudioSpec for SDL_OpenAudio() in (orig).
|
|
* Fills in a sanitized copy in (prepared).
|
|
* Returns non-zero if okay, zero on fatal parameters in (orig).
|
|
*/
|
|
static int
|
|
prepare_audiospec(const SDL_AudioSpec * orig, SDL_AudioSpec * prepared)
|
|
{
|
|
SDL_memcpy(prepared, orig, sizeof(SDL_AudioSpec));
|
|
|
|
if (orig->callback == NULL) {
|
|
SDL_SetError("SDL_OpenAudio() passed a NULL callback");
|
|
return 0;
|
|
}
|
|
|
|
if (orig->freq == 0) {
|
|
const char *env = SDL_getenv("SDL_AUDIO_FREQUENCY");
|
|
if ((!env) || ((prepared->freq = SDL_atoi(env)) == 0)) {
|
|
prepared->freq = 22050; /* a reasonable default */
|
|
}
|
|
}
|
|
|
|
if (orig->format == 0) {
|
|
const char *env = SDL_getenv("SDL_AUDIO_FORMAT");
|
|
if ((!env) || ((prepared->format = SDL_ParseAudioFormat(env)) == 0)) {
|
|
prepared->format = AUDIO_S16; /* a reasonable default */
|
|
}
|
|
}
|
|
|
|
switch (orig->channels) {
|
|
case 0:{
|
|
const char *env = SDL_getenv("SDL_AUDIO_CHANNELS");
|
|
if ((!env) || ((prepared->channels = (Uint8) SDL_atoi(env)) == 0)) {
|
|
prepared->channels = 2; /* a reasonable default */
|
|
}
|
|
break;
|
|
}
|
|
case 1: /* Mono */
|
|
case 2: /* Stereo */
|
|
case 4: /* surround */
|
|
case 6: /* surround with center and lfe */
|
|
break;
|
|
default:
|
|
SDL_SetError("Unsupported number of audio channels.");
|
|
return 0;
|
|
}
|
|
|
|
if (orig->samples == 0) {
|
|
const char *env = SDL_getenv("SDL_AUDIO_SAMPLES");
|
|
if ((!env) || ((prepared->samples = (Uint16) SDL_atoi(env)) == 0)) {
|
|
/* Pick a default of ~46 ms at desired frequency */
|
|
/* !!! FIXME: remove this when the non-Po2 resampling is in. */
|
|
const int samples = (prepared->freq / 1000) * 46;
|
|
int power2 = 1;
|
|
while (power2 < samples) {
|
|
power2 *= 2;
|
|
}
|
|
prepared->samples = power2;
|
|
}
|
|
}
|
|
|
|
/* Calculate the silence and size of the audio specification */
|
|
SDL_CalculateAudioSpec(prepared);
|
|
|
|
return 1;
|
|
}
|
|
|
|
|
|
static SDL_AudioDeviceID
|
|
open_audio_device(const char *devname, int iscapture,
|
|
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
|
|
int allowed_changes, int min_id)
|
|
{
|
|
SDL_AudioDeviceID id = 0;
|
|
SDL_AudioSpec _obtained;
|
|
SDL_AudioDevice *device;
|
|
SDL_bool build_cvt;
|
|
int i = 0;
|
|
|
|
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
|
|
SDL_SetError("Audio subsystem is not initialized");
|
|
return 0;
|
|
}
|
|
|
|
if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
|
|
SDL_SetError("No capture support");
|
|
return 0;
|
|
}
|
|
|
|
if (!obtained) {
|
|
obtained = &_obtained;
|
|
}
|
|
if (!prepare_audiospec(desired, obtained)) {
|
|
return 0;
|
|
}
|
|
|
|
/* If app doesn't care about a specific device, let the user override. */
|
|
if (devname == NULL) {
|
|
devname = SDL_getenv("SDL_AUDIO_DEVICE_NAME");
|
|
}
|
|
|
|
/*
|
|
* Catch device names at the high level for the simple case...
|
|
* This lets us have a basic "device enumeration" for systems that
|
|
* don't have multiple devices, but makes sure the device name is
|
|
* always NULL when it hits the low level.
|
|
*
|
|
* Also make sure that the simple case prevents multiple simultaneous
|
|
* opens of the default system device.
|
|
*/
|
|
|
|
if ((iscapture) && (current_audio.impl.OnlyHasDefaultInputDevice)) {
|
|
if ((devname) && (SDL_strcmp(devname, DEFAULT_INPUT_DEVNAME) != 0)) {
|
|
SDL_SetError("No such device");
|
|
return 0;
|
|
}
|
|
devname = NULL;
|
|
|
|
for (i = 0; i < SDL_arraysize(open_devices); i++) {
|
|
if ((open_devices[i]) && (open_devices[i]->iscapture)) {
|
|
SDL_SetError("Audio device already open");
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
if ((!iscapture) && (current_audio.impl.OnlyHasDefaultOutputDevice)) {
|
|
if ((devname) && (SDL_strcmp(devname, DEFAULT_OUTPUT_DEVNAME) != 0)) {
|
|
SDL_SetError("No such device");
|
|
return 0;
|
|
}
|
|
devname = NULL;
|
|
|
|
for (i = 0; i < SDL_arraysize(open_devices); i++) {
|
|
if ((open_devices[i]) && (!open_devices[i]->iscapture)) {
|
|
SDL_SetError("Audio device already open");
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
device = (SDL_AudioDevice *) SDL_AllocAudioMem(sizeof(SDL_AudioDevice));
|
|
if (device == NULL) {
|
|
SDL_OutOfMemory();
|
|
return 0;
|
|
}
|
|
SDL_memset(device, '\0', sizeof(SDL_AudioDevice));
|
|
device->spec = *obtained;
|
|
device->enabled = 1;
|
|
device->paused = 1;
|
|
device->iscapture = iscapture;
|
|
|
|
/* Create a semaphore for locking the sound buffers */
|
|
if (!current_audio.impl.SkipMixerLock) {
|
|
device->mixer_lock = SDL_CreateMutex();
|
|
if (device->mixer_lock == NULL) {
|
|
close_audio_device(device);
|
|
SDL_SetError("Couldn't create mixer lock");
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (!current_audio.impl.OpenDevice(device, devname, iscapture)) {
|
|
close_audio_device(device);
|
|
return 0;
|
|
}
|
|
device->opened = 1;
|
|
|
|
/* Allocate a fake audio memory buffer */
|
|
device->fake_stream = SDL_AllocAudioMem(device->spec.size);
|
|
if (device->fake_stream == NULL) {
|
|
close_audio_device(device);
|
|
SDL_OutOfMemory();
|
|
return 0;
|
|
}
|
|
|
|
/* If the audio driver changes the buffer size, accept it */
|
|
if (device->spec.samples != obtained->samples) {
|
|
obtained->samples = device->spec.samples;
|
|
SDL_CalculateAudioSpec(obtained);
|
|
}
|
|
|
|
/* See if we need to do any conversion */
|
|
build_cvt = SDL_FALSE;
|
|
if (obtained->freq != device->spec.freq) {
|
|
if (allowed_changes & SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) {
|
|
obtained->freq = device->spec.freq;
|
|
} else {
|
|
build_cvt = SDL_TRUE;
|
|
}
|
|
}
|
|
if (obtained->format != device->spec.format) {
|
|
if (allowed_changes & SDL_AUDIO_ALLOW_FORMAT_CHANGE) {
|
|
obtained->format = device->spec.format;
|
|
} else {
|
|
build_cvt = SDL_TRUE;
|
|
}
|
|
}
|
|
if (obtained->channels != device->spec.channels) {
|
|
if (allowed_changes & SDL_AUDIO_ALLOW_CHANNELS_CHANGE) {
|
|
obtained->channels = device->spec.channels;
|
|
} else {
|
|
build_cvt = SDL_TRUE;
|
|
}
|
|
}
|
|
if (build_cvt) {
|
|
/* Build an audio conversion block */
|
|
if (SDL_BuildAudioCVT(&device->convert,
|
|
obtained->format, obtained->channels,
|
|
obtained->freq,
|
|
device->spec.format, device->spec.channels,
|
|
device->spec.freq) < 0) {
|
|
close_audio_device(device);
|
|
return 0;
|
|
}
|
|
if (device->convert.needed) {
|
|
device->convert.len = (int) (((double) obtained->size) /
|
|
device->convert.len_ratio);
|
|
|
|
device->convert.buf =
|
|
(Uint8 *) SDL_AllocAudioMem(device->convert.len *
|
|
device->convert.len_mult);
|
|
if (device->convert.buf == NULL) {
|
|
close_audio_device(device);
|
|
SDL_OutOfMemory();
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Find an available device ID and store the structure... */
|
|
for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
|
|
if (open_devices[id] == NULL) {
|
|
open_devices[id] = device;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (id == SDL_arraysize(open_devices)) {
|
|
SDL_SetError("Too many open audio devices");
|
|
close_audio_device(device);
|
|
return 0;
|
|
}
|
|
|
|
/* Start the audio thread if necessary */
|
|
if (!current_audio.impl.ProvidesOwnCallbackThread) {
|
|
/* Start the audio thread */
|
|
/* !!! FIXME: this is nasty. */
|
|
#if (defined(__WIN32__) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC)
|
|
#undef SDL_CreateThread
|
|
device->thread = SDL_CreateThread(SDL_RunAudio, device, NULL, NULL);
|
|
#else
|
|
device->thread = SDL_CreateThread(SDL_RunAudio, device);
|
|
#endif
|
|
if (device->thread == NULL) {
|
|
SDL_CloseAudioDevice(id + 1);
|
|
SDL_SetError("Couldn't create audio thread");
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
return id + 1;
|
|
}
|
|
|
|
|
|
int
|
|
SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
|
|
{
|
|
SDL_AudioDeviceID id = 0;
|
|
|
|
/* Start up the audio driver, if necessary. This is legacy behaviour! */
|
|
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
|
|
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
|
|
return (-1);
|
|
}
|
|
}
|
|
|
|
/* SDL_OpenAudio() is legacy and can only act on Device ID #1. */
|
|
if (open_devices[0] != NULL) {
|
|
SDL_SetError("Audio device is already opened");
|
|
return (-1);
|
|
}
|
|
|
|
if (obtained) {
|
|
id = open_audio_device(NULL, 0, desired, obtained,
|
|
SDL_AUDIO_ALLOW_ANY_CHANGE, 1);
|
|
} else {
|
|
id = open_audio_device(NULL, 0, desired, desired, 0, 1);
|
|
}
|
|
if (id > 1) { /* this should never happen in theory... */
|
|
SDL_CloseAudioDevice(id);
|
|
SDL_SetError("Internal error"); /* MUST be Device ID #1! */
|
|
return (-1);
|
|
}
|
|
|
|
return ((id == 0) ? -1 : 0);
|
|
}
|
|
|
|
SDL_AudioDeviceID
|
|
SDL_OpenAudioDevice(const char *device, int iscapture,
|
|
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
|
|
int allowed_changes)
|
|
{
|
|
return open_audio_device(device, iscapture, desired, obtained,
|
|
allowed_changes, 2);
|
|
}
|
|
|
|
SDL_audiostatus
|
|
SDL_GetAudioDeviceStatus(SDL_AudioDeviceID devid)
|
|
{
|
|
SDL_AudioDevice *device = get_audio_device(devid);
|
|
SDL_audiostatus status = SDL_AUDIO_STOPPED;
|
|
if (device && device->enabled) {
|
|
if (device->paused) {
|
|
status = SDL_AUDIO_PAUSED;
|
|
} else {
|
|
status = SDL_AUDIO_PLAYING;
|
|
}
|
|
}
|
|
return (status);
|
|
}
|
|
|
|
|
|
SDL_audiostatus
|
|
SDL_GetAudioStatus(void)
|
|
{
|
|
return SDL_GetAudioDeviceStatus(1);
|
|
}
|
|
|
|
void
|
|
SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
|
|
{
|
|
SDL_AudioDevice *device = get_audio_device(devid);
|
|
if (device) {
|
|
device->paused = pause_on;
|
|
}
|
|
}
|
|
|
|
void
|
|
SDL_PauseAudio(int pause_on)
|
|
{
|
|
SDL_PauseAudioDevice(1, pause_on);
|
|
}
|
|
|
|
|
|
void
|
|
SDL_LockAudioDevice(SDL_AudioDeviceID devid)
|
|
{
|
|
/* Obtain a lock on the mixing buffers */
|
|
SDL_AudioDevice *device = get_audio_device(devid);
|
|
if (device) {
|
|
current_audio.impl.LockDevice(device);
|
|
}
|
|
}
|
|
|
|
void
|
|
SDL_LockAudio(void)
|
|
{
|
|
SDL_LockAudioDevice(1);
|
|
}
|
|
|
|
void
|
|
SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
|
|
{
|
|
/* Obtain a lock on the mixing buffers */
|
|
SDL_AudioDevice *device = get_audio_device(devid);
|
|
if (device) {
|
|
current_audio.impl.UnlockDevice(device);
|
|
}
|
|
}
|
|
|
|
void
|
|
SDL_UnlockAudio(void)
|
|
{
|
|
SDL_UnlockAudioDevice(1);
|
|
}
|
|
|
|
void
|
|
SDL_CloseAudioDevice(SDL_AudioDeviceID devid)
|
|
{
|
|
SDL_AudioDevice *device = get_audio_device(devid);
|
|
if (device) {
|
|
close_audio_device(device);
|
|
open_devices[devid - 1] = NULL;
|
|
}
|
|
}
|
|
|
|
void
|
|
SDL_CloseAudio(void)
|
|
{
|
|
SDL_CloseAudioDevice(1);
|
|
}
|
|
|
|
void
|
|
SDL_AudioQuit(void)
|
|
{
|
|
SDL_AudioDeviceID i;
|
|
for (i = 0; i < SDL_arraysize(open_devices); i++) {
|
|
SDL_CloseAudioDevice(i);
|
|
}
|
|
|
|
/* Free the driver data */
|
|
current_audio.impl.Deinitialize();
|
|
SDL_memset(¤t_audio, '\0', sizeof(current_audio));
|
|
SDL_memset(open_devices, '\0', sizeof(open_devices));
|
|
}
|
|
|
|
#define NUM_FORMATS 10
|
|
static int format_idx;
|
|
static int format_idx_sub;
|
|
static SDL_AudioFormat format_list[NUM_FORMATS][NUM_FORMATS] = {
|
|
{AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
|
|
AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB},
|
|
{AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB,
|
|
AUDIO_U16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB},
|
|
{AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S32LSB,
|
|
AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S32MSB,
|
|
AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_S32LSB,
|
|
AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_S32MSB,
|
|
AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S16LSB,
|
|
AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S16MSB,
|
|
AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_S16LSB,
|
|
AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8},
|
|
{AUDIO_F32MSB, AUDIO_F32LSB, AUDIO_S32MSB, AUDIO_S32LSB, AUDIO_S16MSB,
|
|
AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8},
|
|
};
|
|
|
|
SDL_AudioFormat
|
|
SDL_FirstAudioFormat(SDL_AudioFormat 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());
|
|
}
|
|
|
|
SDL_AudioFormat
|
|
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 = SDL_AUDIO_BITSIZE(spec->format) / 8;
|
|
spec->size *= spec->channels;
|
|
spec->size *= spec->samples;
|
|
}
|
|
|
|
|
|
/*
|
|
* Moved here from SDL_mixer.c, since it relies on internals of an opened
|
|
* audio device (and is deprecated, by the way!).
|
|
*/
|
|
void
|
|
SDL_MixAudio(Uint8 * dst, const Uint8 * src, Uint32 len, int volume)
|
|
{
|
|
/* Mix the user-level audio format */
|
|
SDL_AudioDevice *device = get_audio_device(1);
|
|
if (device != NULL) {
|
|
SDL_AudioFormat format;
|
|
if (device->convert.needed) {
|
|
format = device->convert.src_format;
|
|
} else {
|
|
format = device->spec.format;
|
|
}
|
|
SDL_MixAudioFormat(dst, src, format, len, volume);
|
|
}
|
|
}
|
|
|
|
/* vi: set ts=4 sw=4 expandtab: */
|