mirror of https://github.com/encounter/SDL.git
1456 lines
43 KiB
C
1456 lines
43 KiB
C
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/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2015 Sam Lantinga <slouken@libsdl.org>
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This software is provided 'as-is', without any express or implied
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warranty. In no event will the authors be held liable for any damages
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arising from the use of this software.
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Permission is granted to anyone to use this software for any purpose,
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including commercial applications, and to alter it and redistribute it
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freely, subject to the following restrictions:
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1. The origin of this software must not be misrepresented; you must not
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claim that you wrote the original software. If you use this software
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in a product, an acknowledgment in the product documentation would be
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appreciated but is not required.
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2. Altered source versions must be plainly marked as such, and must not be
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misrepresented as being the original software.
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3. This notice may not be removed or altered from any source distribution.
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*/
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#include "../SDL_internal.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.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 ALSA_bootstrap;
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extern AudioBootStrap PULSEAUDIO_bootstrap;
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extern AudioBootStrap QSAAUDIO_bootstrap;
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extern AudioBootStrap SUNAUDIO_bootstrap;
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extern AudioBootStrap ARTS_bootstrap;
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extern AudioBootStrap ESD_bootstrap;
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extern AudioBootStrap NACLAUD_bootstrap;
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extern AudioBootStrap NAS_bootstrap;
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extern AudioBootStrap XAUDIO2_bootstrap;
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extern AudioBootStrap DSOUND_bootstrap;
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extern AudioBootStrap WINMM_bootstrap;
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extern AudioBootStrap PAUDIO_bootstrap;
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extern AudioBootStrap HAIKUAUDIO_bootstrap;
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extern AudioBootStrap COREAUDIO_bootstrap;
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extern AudioBootStrap SNDMGR_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 DART_bootstrap;
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extern AudioBootStrap NDSAUD_bootstrap;
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extern AudioBootStrap FUSIONSOUND_bootstrap;
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extern AudioBootStrap ANDROIDAUD_bootstrap;
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extern AudioBootStrap PSPAUD_bootstrap;
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extern AudioBootStrap SNDIO_bootstrap;
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extern AudioBootStrap EmscriptenAudio_bootstrap;
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/* Available audio drivers */
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static const AudioBootStrap *const bootstrap[] = {
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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_SNDIO
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&SNDIO_bootstrap,
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#endif
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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_OSS
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&DSP_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_QSA
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&QSAAUDIO_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_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_NACL
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&NACLAUD_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_XAUDIO2
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&XAUDIO2_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_WINMM
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&WINMM_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_HAIKU
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&HAIKUAUDIO_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_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_FUSIONSOUND
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&FUSIONSOUND_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_ANDROID
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&ANDROIDAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_PSP
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&PSPAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_EMSCRIPTEN
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&EmscriptenAudio_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 void
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SDL_AudioDetectDevices_Default(void)
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{
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/* you have to write your own implementation if these assertions fail. */
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SDL_assert(current_audio.impl.OnlyHasDefaultOutputDevice);
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SDL_assert(current_audio.impl.OnlyHasDefaultInputDevice || !current_audio.impl.HasCaptureSupport);
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SDL_AddAudioDevice(SDL_FALSE, DEFAULT_OUTPUT_DEVNAME, (void *) ((size_t) 0x1));
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if (current_audio.impl.HasCaptureSupport) {
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SDL_AddAudioDevice(SDL_TRUE, DEFAULT_INPUT_DEVNAME, (void *) ((size_t) 0x2));
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}
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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 int
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SDL_AudioGetPendingBytes_Default(_THIS)
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{
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return 0;
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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 void
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SDL_AudioFreeDeviceHandle_Default(void *handle)
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{ /* no-op. */
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}
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static int
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SDL_AudioOpenDevice_Default(_THIS, void *handle, const char *devname, int iscapture)
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{
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return SDL_Unsupported();
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}
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static SDL_INLINE SDL_bool
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is_in_audio_device_thread(SDL_AudioDevice * device)
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{
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/* The device thread locks the same mutex, but not through the public API.
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This check is in case the application, in the audio callback,
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tries to lock the thread that we've already locked from the
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device thread...just in case we only have non-recursive mutexes. */
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if (device->thread && (SDL_ThreadID() == device->threadid)) {
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return SDL_TRUE;
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}
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return SDL_FALSE;
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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 (!is_in_audio_device_thread(device)) {
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SDL_LockMutex(device->mixer_lock);
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}
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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 (!is_in_audio_device_thread(device)) {
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SDL_UnlockMutex(device->mixer_lock);
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}
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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(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(GetPendingBytes);
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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(FreeDeviceHandle);
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FILL_STUB(Deinitialize);
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#undef FILL_STUB
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}
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/* device hotplug support... */
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static int
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add_audio_device(const char *name, void *handle, SDL_AudioDeviceItem **devices, int *devCount)
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{
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int retval = -1;
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const size_t size = sizeof (SDL_AudioDeviceItem) + SDL_strlen(name) + 1;
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SDL_AudioDeviceItem *item = (SDL_AudioDeviceItem *) SDL_malloc(size);
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if (item == NULL) {
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return -1;
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}
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SDL_assert(handle != NULL); /* we reserve NULL, audio backends can't use it. */
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item->handle = handle;
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SDL_strlcpy(item->name, name, size - sizeof (SDL_AudioDeviceItem));
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SDL_LockMutex(current_audio.detectionLock);
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item->next = *devices;
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*devices = item;
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retval = (*devCount)++;
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SDL_UnlockMutex(current_audio.detectionLock);
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return retval;
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}
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static SDL_INLINE int
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add_capture_device(const char *name, void *handle)
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{
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/* !!! FIXME: add this later. SDL_assert(current_audio.impl.HasCaptureSupport);*/
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return add_audio_device(name, handle, ¤t_audio.inputDevices, ¤t_audio.inputDeviceCount);
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}
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static SDL_INLINE int
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add_output_device(const char *name, void *handle)
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{
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return add_audio_device(name, handle, ¤t_audio.outputDevices, ¤t_audio.outputDeviceCount);
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}
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static void
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free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
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{
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SDL_AudioDeviceItem *item, *next;
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for (item = *devices; item != NULL; item = next) {
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next = item->next;
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if (item->handle != NULL) {
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current_audio.impl.FreeDeviceHandle(item->handle);
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}
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SDL_free(item);
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}
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*devices = NULL;
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*devCount = 0;
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}
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/* The audio backends call this when a new device is plugged in. */
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void
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SDL_AddAudioDevice(const int iscapture, const char *name, void *handle)
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{
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const int device_index = iscapture ? add_capture_device(name, handle) : add_output_device(name, handle);
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if (device_index != -1) {
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/* Post the event, if desired */
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if (SDL_GetEventState(SDL_AUDIODEVICEADDED) == SDL_ENABLE) {
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SDL_Event event;
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SDL_zero(event);
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event.adevice.type = SDL_AUDIODEVICEADDED;
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event.adevice.which = device_index;
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event.adevice.iscapture = iscapture;
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SDL_PushEvent(&event);
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}
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}
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}
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/* The audio backends call this when a currently-opened device is lost. */
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void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device)
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{
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SDL_assert(get_audio_device(device->id) == device);
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if (!device->enabled) {
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return;
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}
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/* Ends the audio callback and mark the device as STOPPED, but the
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app still needs to close the device to free resources. */
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current_audio.impl.LockDevice(device);
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device->enabled = 0;
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current_audio.impl.UnlockDevice(device);
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/* Post the event, if desired */
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if (SDL_GetEventState(SDL_AUDIODEVICEREMOVED) == SDL_ENABLE) {
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SDL_Event event;
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SDL_zero(event);
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event.adevice.type = SDL_AUDIODEVICEREMOVED;
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event.adevice.which = device->id;
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event.adevice.iscapture = device->iscapture ? 1 : 0;
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SDL_PushEvent(&event);
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}
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}
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static void
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mark_device_removed(void *handle, SDL_AudioDeviceItem *devices, SDL_bool *removedFlag)
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{
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SDL_AudioDeviceItem *item;
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SDL_assert(handle != NULL);
|
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for (item = devices; item != NULL; item = item->next) {
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if (item->handle == handle) {
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item->handle = NULL;
|
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*removedFlag = SDL_TRUE;
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return;
|
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}
|
||
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}
|
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}
|
||
|
|
||
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/* The audio backends call this when a device is removed from the system. */
|
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|
void
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SDL_RemoveAudioDevice(const int iscapture, void *handle)
|
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|
{
|
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SDL_LockMutex(current_audio.detectionLock);
|
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|
if (iscapture) {
|
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mark_device_removed(handle, current_audio.inputDevices, ¤t_audio.captureDevicesRemoved);
|
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|
} else {
|
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mark_device_removed(handle, current_audio.outputDevices, ¤t_audio.outputDevicesRemoved);
|
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|
}
|
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SDL_UnlockMutex(current_audio.detectionLock);
|
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current_audio.impl.FreeDeviceHandle(handle);
|
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|
}
|
||
|
|
||
|
|
||
|
|
||
|
/* buffer queueing support... */
|
||
|
|
||
|
/* this expects that you managed thread safety elsewhere. */
|
||
|
static void
|
||
|
free_audio_queue(SDL_AudioBufferQueue *buffer)
|
||
|
{
|
||
|
while (buffer) {
|
||
|
SDL_AudioBufferQueue *next = buffer->next;
|
||
|
SDL_free(buffer);
|
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|
buffer = next;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
static void SDLCALL
|
||
|
SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int _len)
|
||
|
{
|
||
|
/* this function always holds the mixer lock before being called. */
|
||
|
Uint32 len = (Uint32) _len;
|
||
|
SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
|
||
|
SDL_AudioBufferQueue *buffer;
|
||
|
|
||
|
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
|
||
|
SDL_assert(_len >= 0); /* this shouldn't ever happen, right?! */
|
||
|
|
||
|
while ((len > 0) && ((buffer = device->buffer_queue_head) != NULL)) {
|
||
|
const Uint32 avail = buffer->datalen - buffer->startpos;
|
||
|
const Uint32 cpy = SDL_min(len, avail);
|
||
|
SDL_assert(device->queued_bytes >= avail);
|
||
|
|
||
|
SDL_memcpy(stream, buffer->data + buffer->startpos, cpy);
|
||
|
buffer->startpos += cpy;
|
||
|
stream += cpy;
|
||
|
device->queued_bytes -= cpy;
|
||
|
len -= cpy;
|
||
|
|
||
|
if (buffer->startpos == buffer->datalen) { /* packet is done, put it in the pool. */
|
||
|
device->buffer_queue_head = buffer->next;
|
||
|
SDL_assert((buffer->next != NULL) || (buffer == device->buffer_queue_tail));
|
||
|
buffer->next = device->buffer_queue_pool;
|
||
|
device->buffer_queue_pool = buffer;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
|
||
|
|
||
|
if (len > 0) { /* fill any remaining space in the stream with silence. */
|
||
|
SDL_assert(device->buffer_queue_head == NULL);
|
||
|
SDL_memset(stream, device->spec.silence, len);
|
||
|
}
|
||
|
|
||
|
if (device->buffer_queue_head == NULL) {
|
||
|
device->buffer_queue_tail = NULL; /* in case we drained the queue entirely. */
|
||
|
}
|
||
|
}
|
||
|
|
||
|
int
|
||
|
SDL_QueueAudio(SDL_AudioDeviceID devid, const void *_data, Uint32 len)
|
||
|
{
|
||
|
SDL_AudioDevice *device = get_audio_device(devid);
|
||
|
const Uint8 *data = (const Uint8 *) _data;
|
||
|
SDL_AudioBufferQueue *orighead;
|
||
|
SDL_AudioBufferQueue *origtail;
|
||
|
Uint32 origlen;
|
||
|
Uint32 datalen;
|
||
|
|
||
|
if (!device) {
|
||
|
return -1; /* get_audio_device() will have set the error state */
|
||
|
}
|
||
|
|
||
|
if (device->spec.callback != SDL_BufferQueueDrainCallback) {
|
||
|
return SDL_SetError("Audio device has a callback, queueing not allowed");
|
||
|
}
|
||
|
|
||
|
current_audio.impl.LockDevice(device);
|
||
|
|
||
|
orighead = device->buffer_queue_head;
|
||
|
origtail = device->buffer_queue_tail;
|
||
|
origlen = origtail ? origtail->datalen : 0;
|
||
|
|
||
|
while (len > 0) {
|
||
|
SDL_AudioBufferQueue *packet = device->buffer_queue_tail;
|
||
|
SDL_assert(!packet || (packet->datalen <= SDL_AUDIOBUFFERQUEUE_PACKETLEN));
|
||
|
if (!packet || (packet->datalen >= SDL_AUDIOBUFFERQUEUE_PACKETLEN)) {
|
||
|
/* tail packet missing or completely full; we need a new packet. */
|
||
|
packet = device->buffer_queue_pool;
|
||
|
if (packet != NULL) {
|
||
|
/* we have one available in the pool. */
|
||
|
device->buffer_queue_pool = packet->next;
|
||
|
} else {
|
||
|
/* Have to allocate a new one! */
|
||
|
packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
|
||
|
if (packet == NULL) {
|
||
|
/* uhoh, reset so we've queued nothing new, free what we can. */
|
||
|
if (!origtail) {
|
||
|
packet = device->buffer_queue_head; /* whole queue. */
|
||
|
} else {
|
||
|
packet = origtail->next; /* what we added to existing queue. */
|
||
|
origtail->next = NULL;
|
||
|
origtail->datalen = origlen;
|
||
|
}
|
||
|
device->buffer_queue_head = orighead;
|
||
|
device->buffer_queue_tail = origtail;
|
||
|
device->buffer_queue_pool = NULL;
|
||
|
|
||
|
current_audio.impl.UnlockDevice(device);
|
||
|
|
||
|
free_audio_queue(packet); /* give back what we can. */
|
||
|
|
||
|
return SDL_OutOfMemory();
|
||
|
}
|
||
|
}
|
||
|
packet->datalen = 0;
|
||
|
packet->startpos = 0;
|
||
|
packet->next = NULL;
|
||
|
|
||
|
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
|
||
|
if (device->buffer_queue_tail == NULL) {
|
||
|
device->buffer_queue_head = packet;
|
||
|
} else {
|
||
|
device->buffer_queue_tail->next = packet;
|
||
|
}
|
||
|
device->buffer_queue_tail = packet;
|
||
|
}
|
||
|
|
||
|
datalen = SDL_min(len, SDL_AUDIOBUFFERQUEUE_PACKETLEN - packet->datalen);
|
||
|
SDL_memcpy(packet->data + packet->datalen, data, datalen);
|
||
|
data += datalen;
|
||
|
len -= datalen;
|
||
|
packet->datalen += datalen;
|
||
|
device->queued_bytes += datalen;
|
||
|
}
|
||
|
|
||
|
current_audio.impl.UnlockDevice(device);
|
||
|
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
Uint32
|
||
|
SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
|
||
|
{
|
||
|
Uint32 retval = 0;
|
||
|
SDL_AudioDevice *device = get_audio_device(devid);
|
||
|
|
||
|
/* Nothing to do unless we're set up for queueing. */
|
||
|
if (device && (device->spec.callback == SDL_BufferQueueDrainCallback)) {
|
||
|
current_audio.impl.LockDevice(device);
|
||
|
retval = device->queued_bytes + current_audio.impl.GetPendingBytes(device);
|
||
|
current_audio.impl.UnlockDevice(device);
|
||
|
}
|
||
|
|
||
|
return retval;
|
||
|
}
|
||
|
|
||
|
void
|
||
|
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
|
||
|
{
|
||
|
SDL_AudioDevice *device = get_audio_device(devid);
|
||
|
SDL_AudioBufferQueue *buffer = NULL;
|
||
|
if (!device) {
|
||
|
return; /* nothing to do. */
|
||
|
}
|
||
|
|
||
|
/* Blank out the device and release the mutex. Free it afterwards. */
|
||
|
current_audio.impl.LockDevice(device);
|
||
|
buffer = device->buffer_queue_head;
|
||
|
device->buffer_queue_tail = NULL;
|
||
|
device->buffer_queue_head = NULL;
|
||
|
device->queued_bytes = 0;
|
||
|
current_audio.impl.UnlockDevice(device);
|
||
|
|
||
|
free_audio_queue(buffer);
|
||
|
}
|
||
|
|
||
|
|
||
|
/* The general mixing thread function */
|
||
|
int SDLCALL
|
||
|
SDL_RunAudio(void *devicep)
|
||
|
{
|
||
|
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
|
||
|
const int silence = (int) device->spec.silence;
|
||
|
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
|
||
|
const int stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
|
||
|
Uint8 *stream;
|
||
|
void *udata = device->spec.userdata;
|
||
|
void (SDLCALL *fill) (void *, Uint8 *, int) = device->spec.callback;
|
||
|
|
||
|
/* The audio mixing is always a high priority thread */
|
||
|
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
|
||
|
|
||
|
/* Perform any thread setup */
|
||
|
device->threadid = SDL_ThreadID();
|
||
|
current_audio.impl.ThreadInit(device);
|
||
|
|
||
|
/* Loop, filling the audio buffers */
|
||
|
while (!device->shutdown) {
|
||
|
/* Fill the current buffer with sound */
|
||
|
if (device->convert.needed) {
|
||
|
stream = device->convert.buf;
|
||
|
} else if (device->enabled) {
|
||
|
stream = current_audio.impl.GetDeviceBuf(device);
|
||
|
} else {
|
||
|
/* if the device isn't enabled, we still write to the
|
||
|
fake_stream, so the app's callback will fire with
|
||
|
a regular frequency, in case they depend on that
|
||
|
for timing or progress. They can use hotplug
|
||
|
now to know if the device failed. */
|
||
|
stream = NULL;
|
||
|
}
|
||
|
|
||
|
if (stream == NULL) {
|
||
|
stream = device->fake_stream;
|
||
|
}
|
||
|
|
||
|
/* !!! FIXME: this should be LockDevice. */
|
||
|
SDL_LockMutex(device->mixer_lock);
|
||
|
if (device->paused) {
|
||
|
SDL_memset(stream, silence, stream_len);
|
||
|
} else {
|
||
|
(*fill) (udata, stream, stream_len);
|
||
|
}
|
||
|
SDL_UnlockMutex(device->mixer_lock);
|
||
|
|
||
|
/* Convert the audio if necessary */
|
||
|
if (device->enabled && device->convert.needed) {
|
||
|
SDL_ConvertAudio(&device->convert);
|
||
|
stream = current_audio.impl.GetDeviceBuf(device);
|
||
|
if (stream == NULL) {
|
||
|
stream = device->fake_stream;
|
||
|
} else {
|
||
|
SDL_memcpy(stream, device->convert.buf,
|
||
|
device->convert.len_cvt);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* Ready current buffer for play and change current buffer */
|
||
|
if (stream == device->fake_stream) {
|
||
|
SDL_Delay(delay);
|
||
|
} else {
|
||
|
current_audio.impl.PlayDevice(device);
|
||
|
current_audio.impl.WaitDevice(device);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* Wait for the audio to drain. */
|
||
|
current_audio.impl.WaitDone(device);
|
||
|
|
||
|
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;
|
||
|
|
||
|
if (SDL_WasInit(SDL_INIT_AUDIO)) {
|
||
|
SDL_AudioQuit(); /* shutdown driver if already running. */
|
||
|
}
|
||
|
|
||
|
SDL_zero(current_audio);
|
||
|
SDL_zero(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_strncasecmp(backend->name, driver_name, SDL_strlen(driver_name)) != 0)) ||
|
||
|
(!driver_name && backend->demand_only)) {
|
||
|
continue;
|
||
|
}
|
||
|
|
||
|
tried_to_init = 1;
|
||
|
SDL_zero(current_audio);
|
||
|
current_audio.name = backend->name;
|
||
|
current_audio.desc = backend->desc;
|
||
|
initialized = backend->init(¤t_audio.impl);
|
||
|
}
|
||
|
|
||
|
if (!initialized) {
|
||
|
/* specific drivers will set the error message if they fail... */
|
||
|
if (!tried_to_init) {
|
||
|
if (driver_name) {
|
||
|
SDL_SetError("Audio target '%s' not available", driver_name);
|
||
|
} else {
|
||
|
SDL_SetError("No available audio device");
|
||
|
}
|
||
|
}
|
||
|
|
||
|
SDL_zero(current_audio);
|
||
|
return -1; /* No driver was available, so fail. */
|
||
|
}
|
||
|
|
||
|
current_audio.detectionLock = SDL_CreateMutex();
|
||
|
|
||
|
finalize_audio_entry_points();
|
||
|
|
||
|
/* Make sure we have a list of devices available at startup. */
|
||
|
current_audio.impl.DetectDevices();
|
||
|
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
/*
|
||
|
* Get the current audio driver name
|
||
|
*/
|
||
|
const char *
|
||
|
SDL_GetCurrentAudioDriver()
|
||
|
{
|
||
|
return current_audio.name;
|
||
|
}
|
||
|
|
||
|
/* Clean out devices that we've removed but had to keep around for stability. */
|
||
|
static void
|
||
|
clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *removedFlag)
|
||
|
{
|
||
|
SDL_AudioDeviceItem *item = *devices;
|
||
|
SDL_AudioDeviceItem *prev = NULL;
|
||
|
int total = 0;
|
||
|
|
||
|
while (item) {
|
||
|
SDL_AudioDeviceItem *next = item->next;
|
||
|
if (item->handle != NULL) {
|
||
|
total++;
|
||
|
prev = item;
|
||
|
} else {
|
||
|
if (prev) {
|
||
|
prev->next = next;
|
||
|
} else {
|
||
|
*devices = next;
|
||
|
}
|
||
|
SDL_free(item);
|
||
|
}
|
||
|
item = next;
|
||
|
}
|
||
|
|
||
|
*devCount = total;
|
||
|
*removedFlag = SDL_FALSE;
|
||
|
}
|
||
|
|
||
|
|
||
|
int
|
||
|
SDL_GetNumAudioDevices(int iscapture)
|
||
|
{
|
||
|
int retval = 0;
|
||
|
|
||
|
if (!SDL_WasInit(SDL_INIT_AUDIO)) {
|
||
|
return -1;
|
||
|
}
|
||
|
|
||
|
SDL_LockMutex(current_audio.detectionLock);
|
||
|
if (iscapture && current_audio.captureDevicesRemoved) {
|
||
|
clean_out_device_list(¤t_audio.inputDevices, ¤t_audio.inputDeviceCount, ¤t_audio.captureDevicesRemoved);
|
||
|
}
|
||
|
|
||
|
if (!iscapture && current_audio.outputDevicesRemoved) {
|
||
|
clean_out_device_list(¤t_audio.outputDevices, ¤t_audio.outputDeviceCount, ¤t_audio.outputDevicesRemoved);
|
||
|
current_audio.outputDevicesRemoved = SDL_FALSE;
|
||
|
}
|
||
|
|
||
|
retval = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
|
||
|
SDL_UnlockMutex(current_audio.detectionLock);
|
||
|
|
||
|
return retval;
|
||
|
}
|
||
|
|
||
|
|
||
|
const char *
|
||
|
SDL_GetAudioDeviceName(int index, int iscapture)
|
||
|
{
|
||
|
const char *retval = NULL;
|
||
|
|
||
|
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_AudioDeviceItem *item;
|
||
|
int i;
|
||
|
|
||
|
SDL_LockMutex(current_audio.detectionLock);
|
||
|
item = iscapture ? current_audio.inputDevices : current_audio.outputDevices;
|
||
|
i = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
|
||
|
if (index < i) {
|
||
|
for (i--; i > index; i--, item = item->next) {
|
||
|
SDL_assert(item != NULL);
|
||
|
}
|
||
|
SDL_assert(item != NULL);
|
||
|
retval = item->name;
|
||
|
}
|
||
|
SDL_UnlockMutex(current_audio.detectionLock);
|
||
|
}
|
||
|
|
||
|
if (retval == NULL) {
|
||
|
SDL_SetError("No such device");
|
||
|
}
|
||
|
|
||
|
return retval;
|
||
|
}
|
||
|
|
||
|
|
||
|
static void
|
||
|
close_audio_device(SDL_AudioDevice * device)
|
||
|
{
|
||
|
device->enabled = 0;
|
||
|
device->shutdown = 1;
|
||
|
if (device->thread != NULL) {
|
||
|
SDL_WaitThread(device->thread, NULL);
|
||
|
}
|
||
|
if (device->mixer_lock != NULL) {
|
||
|
SDL_DestroyMutex(device->mixer_lock);
|
||
|
}
|
||
|
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;
|
||
|
}
|
||
|
|
||
|
free_audio_queue(device->buffer_queue_head);
|
||
|
free_audio_queue(device->buffer_queue_pool);
|
||
|
|
||
|
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->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;
|
||
|
void *handle = NULL;
|
||
|
Uint32 stream_len;
|
||
|
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;
|
||
|
}
|
||
|
|
||
|
/* Find an available device ID... */
|
||
|
for (id = min_id - 1; id < SDL_arraysize(open_devices); id++) {
|
||
|
if (open_devices[id] == NULL) {
|
||
|
break;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
if (id == SDL_arraysize(open_devices)) {
|
||
|
SDL_SetError("Too many open audio devices");
|
||
|
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;
|
||
|
}
|
||
|
}
|
||
|
} else 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;
|
||
|
}
|
||
|
}
|
||
|
} else if (devname != NULL) {
|
||
|
/* if the app specifies an exact string, we can pass the backend
|
||
|
an actual device handle thingey, which saves them the effort of
|
||
|
figuring out what device this was (such as, reenumerating
|
||
|
everything again to find the matching human-readable name).
|
||
|
It might still need to open a device based on the string for,
|
||
|
say, a network audio server, but this optimizes some cases. */
|
||
|
SDL_AudioDeviceItem *item;
|
||
|
SDL_LockMutex(current_audio.detectionLock);
|
||
|
for (item = iscapture ? current_audio.inputDevices : current_audio.outputDevices; item; item = item->next) {
|
||
|
if ((item->handle != NULL) && (SDL_strcmp(item->name, devname) == 0)) {
|
||
|
handle = item->handle;
|
||
|
break;
|
||
|
}
|
||
|
}
|
||
|
SDL_UnlockMutex(current_audio.detectionLock);
|
||
|
}
|
||
|
|
||
|
if (!current_audio.impl.AllowsArbitraryDeviceNames) {
|
||
|
/* has to be in our device list, or the default device. */
|
||
|
if ((handle == NULL) && (devname != NULL)) {
|
||
|
SDL_SetError("No such device.");
|
||
|
return 0;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
device = (SDL_AudioDevice *) SDL_AllocAudioMem(sizeof(SDL_AudioDevice));
|
||
|
if (device == NULL) {
|
||
|
SDL_OutOfMemory();
|
||
|
return 0;
|
||
|
}
|
||
|
SDL_zerop(device);
|
||
|
device->id = id + 1;
|
||
|
device->spec = *obtained;
|
||
|
device->enabled = 1;
|
||
|
device->paused = 1;
|
||
|
device->iscapture = iscapture;
|
||
|
|
||
|
/* Create a mutex 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, handle, devname, iscapture) < 0) {
|
||
|
close_audio_device(device);
|
||
|
return 0;
|
||
|
}
|
||
|
device->opened = 1;
|
||
|
|
||
|
/* 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 the audio driver changes the buffer size, accept it.
|
||
|
This needs to be done after the format is modified above,
|
||
|
otherwise it might not have the correct buffer size.
|
||
|
*/
|
||
|
if (device->spec.samples != obtained->samples) {
|
||
|
obtained->samples = device->spec.samples;
|
||
|
SDL_CalculateAudioSpec(obtained);
|
||
|
}
|
||
|
|
||
|
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) device->spec.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;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
/* Allocate a fake audio memory buffer */
|
||
|
stream_len = (device->convert.needed) ? device->convert.len_cvt : 0;
|
||
|
if (device->spec.size > stream_len) {
|
||
|
stream_len = device->spec.size;
|
||
|
}
|
||
|
SDL_assert(stream_len > 0);
|
||
|
device->fake_stream = (Uint8 *)SDL_AllocAudioMem(stream_len);
|
||
|
if (device->fake_stream == NULL) {
|
||
|
close_audio_device(device);
|
||
|
SDL_OutOfMemory();
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
if (device->spec.callback == NULL) { /* use buffer queueing? */
|
||
|
/* pool a few packets to start. Enough for two callbacks. */
|
||
|
const int packetlen = SDL_AUDIOBUFFERQUEUE_PACKETLEN;
|
||
|
const int wantbytes = ((device->convert.needed) ? device->convert.len : device->spec.size) * 2;
|
||
|
const int wantpackets = (wantbytes / packetlen) + ((wantbytes % packetlen) ? packetlen : 0);
|
||
|
for (i = 0; i < wantpackets; i++) {
|
||
|
SDL_AudioBufferQueue *packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
|
||
|
if (packet) { /* don't care if this fails, we'll deal later. */
|
||
|
packet->datalen = 0;
|
||
|
packet->startpos = 0;
|
||
|
packet->next = device->buffer_queue_pool;
|
||
|
device->buffer_queue_pool = packet;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
device->spec.callback = SDL_BufferQueueDrainCallback;
|
||
|
device->spec.userdata = device;
|
||
|
}
|
||
|
|
||
|
/* add it to our list of open devices. */
|
||
|
open_devices[id] = device;
|
||
|
|
||
|
/* Start the audio thread if necessary */
|
||
|
if (!current_audio.impl.ProvidesOwnCallbackThread) {
|
||
|
/* Start the audio thread */
|
||
|
char name[64];
|
||
|
SDL_snprintf(name, sizeof (name), "SDLAudioDev%d", (int) device->id);
|
||
|
/* !!! FIXME: this is nasty. */
|
||
|
#if defined(__WIN32__) && !defined(HAVE_LIBC)
|
||
|
#undef SDL_CreateThread
|
||
|
#if SDL_DYNAMIC_API
|
||
|
device->thread = SDL_CreateThread_REAL(SDL_RunAudio, name, device, NULL, NULL);
|
||
|
#else
|
||
|
device->thread = SDL_CreateThread(SDL_RunAudio, name, device, NULL, NULL);
|
||
|
#endif
|
||
|
#else
|
||
|
device->thread = SDL_CreateThread(SDL_RunAudio, name, device);
|
||
|
#endif
|
||
|
if (device->thread == NULL) {
|
||
|
SDL_CloseAudioDevice(device->id);
|
||
|
SDL_SetError("Couldn't create audio thread");
|
||
|
return 0;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
return device->id;
|
||
|
}
|
||
|
|
||
|
|
||
|
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, NULL, 0, 1);
|
||
|
}
|
||
|
|
||
|
SDL_assert((id == 0) || (id == 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) {
|
||
|
current_audio.impl.LockDevice(device);
|
||
|
device->paused = pause_on;
|
||
|
current_audio.impl.UnlockDevice(device);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
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;
|
||
|
|
||
|
if (!current_audio.name) { /* not initialized?! */
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
for (i = 0; i < SDL_arraysize(open_devices); i++) {
|
||
|
if (open_devices[i] != NULL) {
|
||
|
SDL_CloseAudioDevice(i+1);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
free_device_list(¤t_audio.outputDevices, ¤t_audio.outputDeviceCount);
|
||
|
free_device_list(¤t_audio.inputDevices, ¤t_audio.inputDeviceCount);
|
||
|
|
||
|
/* Free the driver data */
|
||
|
current_audio.impl.Deinitialize();
|
||
|
|
||
|
SDL_DestroyMutex(current_audio.detectionLock);
|
||
|
|
||
|
SDL_zero(current_audio);
|
||
|
SDL_zero(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: */
|