SDL-mirror/src/audio/SDL_audio.c

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2006 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
/* Allow access to a raw mixing buffer */
#include "SDL.h"
#include "SDL_audio.h"
#include "SDL_timer.h"
#include "SDL_error.h"
#include "SDL_string.h"
#include "SDL_audio_c.h"
#include "SDL_audiomem.h"
#include "SDL_sysaudio.h"
#ifdef __OS2__
// We'll need the DosSetPriority() API!
#define INCL_DOSPROCESS
#include <os2.h>
#endif
/* Available audio drivers */
static AudioBootStrap *bootstrap[] = {
#ifdef OPENBSD_AUDIO_SUPPORT
&OPENBSD_AUDIO_bootstrap,
#endif
#ifdef OSS_SUPPORT
&DSP_bootstrap,
&DMA_bootstrap,
#endif
#ifdef ALSA_SUPPORT
&ALSA_bootstrap,
#endif
Date: Sat, 2 Aug 2003 16:22:51 +0300 From: "Mike Gorchak" Subject: New patches for QNX6 Here my patches for the SDL/QNX: QNXSDL.diff - diff to non-QNX related sources: - updated BUGS file, I think QNX6 is now will be officially supported - configure.in - added shared library support for QNX, and removed dependency between the ALSA and QNX6. - SDL_audio.c - added QNX NTO sound bootstrap insted of ALSA's. - SDL_sysaudio.h - the same. - SDL_nto_audio.c - the same. - SDL_video.c - right now, QNX doesn't offer any method to obtain pointers to the OpenGL functions by function name, so they must be hardcoded in library, otherwise OpenGL will not be supported. - testsprite.c - fixed: do not draw vertical red line if we are in non-double-buffered mode. sdlqnxph.tar.gz - archive of the ./src/video/photon/* . Too many changes in code to make diffs :) : + Added stub for support hide/unhide window event + Added full YUV overlays support. + Added window maximize support. + Added mouse wheel events. + Added support for some specific key codes in Unicode mode (like ESC). + Added more checks to the all memory allocation code. + Added SDL_DOUBLEBUF support in all fullscreen modes. + Added fallback to window mode, if desired fullscreen mode is not supported. + Added stub support for the GL_LoadLibrary and GL_GetProcAddress functions. + Added resizable window support without caption. ! Fixed bug in the Ph_EV_EXPOSE event handler, when rectangles to update is 0 and when width or height of the rectangle is 0. ! Fixed bug in the event handler code. Events has not been passed to the window widget handler. ! Fixed codes for Win keys (Super/Hyper/Menu). ! Fixed memory leak, when deallocation palette. ! Fixed palette emulation code bugs. ! Fixed fullscreen and hwsurface handling. ! Fixed CLOSE button bug. First event was passed to the handler, but second terminated the application. Now all events passed to the application correctly. - Removed all printfs in code, now SDL_SetError used instead of them. - Disabled ToggleFullScreen function. README.QNX - updated README.QNX file. Added much more issues. --HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%40664
2003-08-04 00:52:42 +00:00
#ifdef QNXNTOAUDIO_SUPPORT
&QNXNTOAUDIO_bootstrap,
#endif
#ifdef SUNAUDIO_SUPPORT
&SUNAUDIO_bootstrap,
#endif
#ifdef DMEDIA_SUPPORT
&DMEDIA_bootstrap,
#endif
#ifdef ARTSC_SUPPORT
&ARTSC_bootstrap,
#endif
#ifdef ESD_SUPPORT
&ESD_bootstrap,
#endif
#ifdef NAS_SUPPORT
&NAS_bootstrap,
#endif
#ifdef ENABLE_DIRECTX
&DSOUND_bootstrap,
#endif
#ifdef ENABLE_WINDIB
&WAVEOUT_bootstrap,
#endif
#ifdef __BEOS__
&BAUDIO_bootstrap,
#endif
#ifdef MACOSX
&COREAUDIO_bootstrap,
#endif
#if defined(macintosh) || TARGET_API_MAC_CARBON
&SNDMGR_bootstrap,
#endif
#ifdef _AIX
&Paud_bootstrap,
#endif
#ifdef ENABLE_AHI
&AHI_bootstrap,
#endif
#ifdef MMEAUDIO_SUPPORT
&MMEAUDIO_bootstrap,
#endif
#ifdef MINTAUDIO_SUPPORT
&MINTAUDIO_GSXB_bootstrap,
&MINTAUDIO_MCSN_bootstrap,
&MINTAUDIO_STFA_bootstrap,
&MINTAUDIO_XBIOS_bootstrap,
&MINTAUDIO_DMA8_bootstrap,
#endif
#ifdef DISKAUD_SUPPORT
&DISKAUD_bootstrap,
#endif
#ifdef ENABLE_DC
&DCAUD_bootstrap,
#endif
#ifdef DRENDERER_SUPPORT
&DRENDERER_bootstrap,
#endif
#ifdef __OS2__
&DART_bootstrap,
#endif
NULL
};
SDL_AudioDevice *current_audio = NULL;
/* Various local functions */
int SDL_AudioInit(const char *driver_name);
void SDL_AudioQuit(void);
#ifdef ENABLE_AHI
static int audio_configured = 0;
#endif
/* The general mixing thread function */
int SDL_RunAudio(void *audiop)
{
SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop;
Uint8 *stream;
int stream_len;
void *udata;
void (*fill)(void *userdata,Uint8 *stream, int len);
int silence;
#ifdef ENABLE_AHI
int started = 0;
/* AmigaOS NEEDS that the audio driver is opened in the thread that uses it! */
D(bug("Task audio started audio struct:<%lx>...\n",audiop));
D(bug("Before Openaudio..."));
if(audio->OpenAudio(audio, &audio->spec)==-1)
{
D(bug("Open audio failed...\n"));
return(-1);
}
D(bug("OpenAudio...OK\n"));
#endif
/* Perform any thread setup */
if ( audio->ThreadInit ) {
audio->ThreadInit(audio);
}
audio->threadid = SDL_ThreadID();
/* Set up the mixing function */
fill = audio->spec.callback;
udata = audio->spec.userdata;
#ifdef ENABLE_AHI
audio_configured = 1;
D(bug("Audio configured... Checking for conversion\n"));
SDL_mutexP(audio->mixer_lock);
D(bug("Semaphore obtained...\n"));
#endif
if ( audio->convert.needed ) {
if ( audio->convert.src_format == AUDIO_U8 ) {
silence = 0x80;
} else {
silence = 0;
}
stream_len = audio->convert.len;
} else {
silence = audio->spec.silence;
stream_len = audio->spec.size;
}
stream = audio->fake_stream;
#ifdef ENABLE_AHI
SDL_mutexV(audio->mixer_lock);
D(bug("Entering audio loop...\n"));
#endif
#ifdef __OS2__
// Increase the priority of this thread to make sure that
// the audio will be continuous all the time!
#ifdef USE_DOSSETPRIORITY
#ifdef DEBUG_BUILD
printf("[SDL_RunAudio] : Setting priority to ForegroundServer+0! (TID%d)\n", SDL_ThreadID());
#endif
DosSetPriority(PRTYS_THREAD, PRTYC_FOREGROUNDSERVER, 0, 0);
#endif
#endif
/* Loop, filling the audio buffers */
while ( audio->enabled ) {
/* Wait for new current buffer to finish playing */
if ( stream == audio->fake_stream ) {
SDL_Delay((audio->spec.samples*1000)/audio->spec.freq);
} else {
#ifdef ENABLE_AHI
if ( started > 1 )
#endif
audio->WaitAudio(audio);
}
/* Fill the current buffer with sound */
if ( audio->convert.needed ) {
if ( audio->convert.buf ) {
stream = audio->convert.buf;
} else {
continue;
}
} else {
stream = audio->GetAudioBuf(audio);
if ( stream == NULL ) {
stream = audio->fake_stream;
}
}
memset(stream, silence, stream_len);
if ( ! audio->paused ) {
SDL_mutexP(audio->mixer_lock);
(*fill)(udata, stream, stream_len);
SDL_mutexV(audio->mixer_lock);
}
/* Convert the audio if necessary */
if ( audio->convert.needed ) {
SDL_ConvertAudio(&audio->convert);
stream = audio->GetAudioBuf(audio);
if ( stream == NULL ) {
stream = audio->fake_stream;
}
memcpy(stream, audio->convert.buf,
audio->convert.len_cvt);
}
/* Ready current buffer for play and change current buffer */
if ( stream != audio->fake_stream ) {
audio->PlayAudio(audio);
#ifdef ENABLE_AHI
/* AmigaOS don't have to wait the first time audio is played! */
started++;
#endif
}
}
/* Wait for the audio to drain.. */
if ( audio->WaitDone ) {
audio->WaitDone(audio);
}
#ifdef ENABLE_AHI
D(bug("WaitAudio...Done\n"));
audio->CloseAudio(audio);
D(bug("CloseAudio..Done, subtask exiting...\n"));
audio_configured = 0;
#endif
#ifdef __OS2__
#ifdef DEBUG_BUILD
printf("[SDL_RunAudio] : Task exiting. (TID%d)\n", SDL_ThreadID());
#endif
#endif
return(0);
}
static void SDL_LockAudio_Default(SDL_AudioDevice *audio)
{
if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
return;
}
SDL_mutexP(audio->mixer_lock);
}
static void SDL_UnlockAudio_Default(SDL_AudioDevice *audio)
{
if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
return;
}
SDL_mutexV(audio->mixer_lock);
}
int SDL_AudioInit(const char *driver_name)
{
SDL_AudioDevice *audio;
int i = 0, idx;
/* Check to make sure we don't overwrite 'current_audio' */
if ( current_audio != NULL ) {
SDL_AudioQuit();
}
/* Select the proper audio driver */
audio = NULL;
idx = 0;
#ifdef unix
if ( (driver_name == NULL) && (getenv("ESPEAKER") != NULL) ) {
/* Ahem, we know that if ESPEAKER is set, user probably wants
to use ESD, but don't start it if it's not already running.
This probably isn't the place to do this, but... Shh! :)
*/
for ( i=0; bootstrap[i]; ++i ) {
if ( strcmp(bootstrap[i]->name, "esd") == 0 ) {
const char *esd_no_spawn;
/* Don't start ESD if it's not running */
esd_no_spawn = getenv("ESD_NO_SPAWN");
if ( esd_no_spawn == NULL ) {
putenv("ESD_NO_SPAWN=1");
}
if ( bootstrap[i]->available() ) {
audio = bootstrap[i]->create(0);
break;
}
#ifdef linux /* No unsetenv() on most platforms */
if ( esd_no_spawn == NULL ) {
unsetenv("ESD_NO_SPAWN");
}
#endif
}
}
}
#endif /* unix */
if ( audio == NULL ) {
if ( driver_name != NULL ) {
#if 0 /* This will be replaced with a better driver selection API */
if ( strrchr(driver_name, ':') != NULL ) {
idx = atoi(strrchr(driver_name, ':')+1);
}
#endif
for ( i=0; bootstrap[i]; ++i ) {
if (strncmp(bootstrap[i]->name, driver_name,
strlen(bootstrap[i]->name)) == 0) {
if ( bootstrap[i]->available() ) {
audio=bootstrap[i]->create(idx);
break;
}
}
}
} else {
for ( i=0; bootstrap[i]; ++i ) {
if ( bootstrap[i]->available() ) {
audio = bootstrap[i]->create(idx);
if ( audio != NULL ) {
break;
}
}
}
}
if ( audio == NULL ) {
SDL_SetError("No available audio device");
#if 0 /* Don't fail SDL_Init() if audio isn't available.
SDL_OpenAudio() will handle it at that point. *sigh*
*/
return(-1);
#endif
}
}
current_audio = audio;
if ( current_audio ) {
current_audio->name = bootstrap[i]->name;
if ( !current_audio->LockAudio && !current_audio->UnlockAudio ) {
current_audio->LockAudio = SDL_LockAudio_Default;
current_audio->UnlockAudio = SDL_UnlockAudio_Default;
}
}
return(0);
}
char *SDL_AudioDriverName(char *namebuf, int maxlen)
{
if ( current_audio != NULL ) {
strncpy(namebuf, current_audio->name, maxlen-1);
namebuf[maxlen-1] = '\0';
return(namebuf);
}
return(NULL);
}
int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained)
{
SDL_AudioDevice *audio;
/* Start up the audio driver, if necessary */
if ( ! current_audio ) {
if ( (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) ||
(current_audio == NULL) ) {
return(-1);
}
}
audio = current_audio;
if (audio->opened) {
SDL_SetError("Audio device is already opened");
return(-1);
}
/* Verify some parameters */
if ( desired->callback == NULL ) {
SDL_SetError("SDL_OpenAudio() passed a NULL callback");
return(-1);
}
switch ( desired->channels ) {
case 1: /* Mono */
case 2: /* Stereo */
Here are patches for SDL12 and SDL_mixer for 4 or 6 channel surround sound on Linux using the Alsa driver. To use them, naturally you need a sound card that will do 4 or 6 channels and probably also a recent version of the Alsa drivers and library. Since the only SDL output driver that knows about surround sound is the Alsa driver, you���ll want to choose it, using: export SDL_AUDIODRIVER=alsa There are no syntactic changes to the programming API. No new library calls, no differences in arguments. There are two semantic changes: (1) For library calls with number of channels as an argument, formerly you could use only 1 or 2 for the number of channels. Now you can also use 4 or 6. (2) The two "left" and "right" arguments to Mix_SetPanning, for the case of 4 or 6 channels, no longer simply control the volumes of the left and right channels. Now the "left" argument is converted to an angle and Mix_SetPosition is called, and the "right" argu- ment is ignored. With two exceptions, so far as I know, the modified SDL12 and SDL_mixer work the same way as the original versions, when opened for 1 or 2 channel output. The two exceptions are bugs which I fixed. Well, the first, anyway, is a bug for sure. When rate conversions up or down by a factor of two are applied (in src/audio/SDL_audiocvt.c), streams with different numbers of channels (that is, mono and stereo) are treated the same way: either each sample is copied or every other sample is omitted. This is ok for mono, but for stereo, it is frames that should be copied or omitted, where by "frame" I mean a portion of the stream containing one sample for each channel. (In the SDL source, confusingly, sometimes frames are called "samples".) So for these rate conversions, stereo streams have to be treated differently, and they are, in my modified version. The other problem that might be characterized as a bug arises when SDL_mixer is passed a multichannel chunk which does not have an integral number of frames. Due to the way the effect_position code loops over frames, when the chunk ends with a partial frame, memory outside the chunk buffer will be accessed. In the case of stereo, it���s possible that because malloc may give more memory than requested, this potential problem never actually causes a segment fault. I don���t know. For 6 channel chunks, I do know, and it does cause segment faults. If SDL_mixer is passed defective chunks and this causes a segment fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not it counts as a bug, it���s easy to protect against, so why not? I added code in mixer.c to discard any partial frame at the end of a chunk. Then what about when SDL or SDL_mixer is opened for 4 or 6 chan- nel output? What happens with the parts of the current library designed for stereo? I don���t know whether I���ve covered all the bases, but I���ve tried: (1) For playing 2 channel waves, or other cases where SDL knows it has to match up a 2 channel source with a 4 or 6 channel output, I���ve added code in SDL_audiocvt.c to make the necessary conversions. (2) For playing midis using timidity, I���ve converted timidity to do 4 or 6 channel output, upon request. (3) For playing mods using mikmod, I put ad hoc code in music.c to convert the stereo output that mikmod produces to 4 or 6 chan- nels. Obviously it would be better to change the mikmod code to mix down into 4 or 6 channels, but I have a hard time following the code in mikmod, so I didn���t do that. (4) For playing mp3s, I put ad hoc code in smpeg to copy channels in the case when 4 or 6 channel output is needed. (5) There seems to be no problem with .ogg files - stereo .oggs can be up converted as .wavs are. (6) The effect_position code in SDL_mixer is now generalized to in- clude the cases of 4 and 6 channel streams. I���ve done a very limited amount of compatibility testing for some of the games using SDL I happen to have. For details, see the file TESTS. I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a couple of 6 channel wave files for testing and a 6 channel ogg file. If you have the right hardware and version of Alsa, you should be able to play the wave files with the Alsa utility aplay (and hear all channels, except maybe lfe, for chan-id.wav, since it���s rather faint). Don���t expect aplay to give good sound, though. There���s something wrong with the current version of aplay. The canyon.ogg file is to test loading of 6 channel oggs. After patching and compiling, you can play it with playmus. (My version of ogg123 will not play it, and I had to patch mplayer to get it to play 6 channel oggs.) Greg Lee <greg@ling.lll.hawaii.edu> Thus, July 1, 2004 --HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%40943
2004-08-21 12:27:02 +00:00
case 4: /* surround */
case 6: /* surround with center and lfe */
break;
default:
SDL_SetError("1 (mono) and 2 (stereo) channels supported");
return(-1);
}
#if defined(macintosh) || (defined(__riscos__) && defined(DISABLE_THREADS))
/* FIXME: Need to implement PPC interrupt asm for SDL_LockAudio() */
#else
#if defined(__MINT__) && !defined(ENABLE_THREADS)
/* Uses interrupt driven audio, without thread */
#else
/* Create a semaphore for locking the sound buffers */
audio->mixer_lock = SDL_CreateMutex();
if ( audio->mixer_lock == NULL ) {
SDL_SetError("Couldn't create mixer lock");
SDL_CloseAudio();
return(-1);
}
#endif /* __MINT__ */
#endif /* macintosh */
/* Calculate the silence and size of the audio specification */
SDL_CalculateAudioSpec(desired);
/* Open the audio subsystem */
memcpy(&audio->spec, desired, sizeof(audio->spec));
audio->convert.needed = 0;
audio->enabled = 1;
audio->paused = 1;
#ifndef ENABLE_AHI
/* AmigaOS opens audio inside the main loop */
audio->opened = audio->OpenAudio(audio, &audio->spec)+1;
if ( ! audio->opened ) {
SDL_CloseAudio();
return(-1);
}
#else
D(bug("Locking semaphore..."));
SDL_mutexP(audio->mixer_lock);
#if (defined(_WIN32) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC)
#undef SDL_CreateThread
audio->thread = SDL_CreateThread(SDL_RunAudio, audio, NULL, NULL);
#else
audio->thread = SDL_CreateThread(SDL_RunAudio, audio);
#endif
D(bug("Created thread...\n"));
if ( audio->thread == NULL ) {
SDL_mutexV(audio->mixer_lock);
SDL_CloseAudio();
SDL_SetError("Couldn't create audio thread");
return(-1);
}
while(!audio_configured)
SDL_Delay(100);
#endif
/* If the audio driver changes the buffer size, accept it */
if ( audio->spec.samples != desired->samples ) {
desired->samples = audio->spec.samples;
SDL_CalculateAudioSpec(desired);
}
/* Allocate a fake audio memory buffer */
audio->fake_stream = SDL_AllocAudioMem(audio->spec.size);
if ( audio->fake_stream == NULL ) {
SDL_CloseAudio();
SDL_OutOfMemory();
return(-1);
}
/* See if we need to do any conversion */
if ( obtained != NULL ) {
memcpy(obtained, &audio->spec, sizeof(audio->spec));
} else if ( desired->freq != audio->spec.freq ||
desired->format != audio->spec.format ||
desired->channels != audio->spec.channels ) {
/* Build an audio conversion block */
if ( SDL_BuildAudioCVT(&audio->convert,
desired->format, desired->channels,
desired->freq,
audio->spec.format, audio->spec.channels,
audio->spec.freq) < 0 ) {
SDL_CloseAudio();
return(-1);
}
if ( audio->convert.needed ) {
audio->convert.len = desired->size;
audio->convert.buf =(Uint8 *)SDL_AllocAudioMem(
audio->convert.len*audio->convert.len_mult);
if ( audio->convert.buf == NULL ) {
SDL_CloseAudio();
SDL_OutOfMemory();
return(-1);
}
}
}
#ifndef ENABLE_AHI
/* Start the audio thread if necessary */
switch (audio->opened) {
case 1:
/* Start the audio thread */
#if (defined(_WIN32) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC)
#undef SDL_CreateThread
audio->thread = SDL_CreateThread(SDL_RunAudio, audio, NULL, NULL);
#else
audio->thread = SDL_CreateThread(SDL_RunAudio, audio);
#endif
if ( audio->thread == NULL ) {
SDL_CloseAudio();
SDL_SetError("Couldn't create audio thread");
return(-1);
}
break;
default:
/* The audio is now playing */
break;
}
#else
SDL_mutexV(audio->mixer_lock);
D(bug("SDL_OpenAudio USCITA...\n"));
#endif
return(0);
}
SDL_audiostatus SDL_GetAudioStatus(void)
{
SDL_AudioDevice *audio = current_audio;
SDL_audiostatus status;
status = SDL_AUDIO_STOPPED;
if ( audio && audio->enabled ) {
if ( audio->paused ) {
status = SDL_AUDIO_PAUSED;
} else {
status = SDL_AUDIO_PLAYING;
}
}
return(status);
}
void SDL_PauseAudio (int pause_on)
{
SDL_AudioDevice *audio = current_audio;
if ( audio ) {
audio->paused = pause_on;
}
}
void SDL_LockAudio (void)
{
SDL_AudioDevice *audio = current_audio;
/* Obtain a lock on the mixing buffers */
if ( audio && audio->LockAudio ) {
audio->LockAudio(audio);
}
}
void SDL_UnlockAudio (void)
{
SDL_AudioDevice *audio = current_audio;
/* Release lock on the mixing buffers */
if ( audio && audio->UnlockAudio ) {
audio->UnlockAudio(audio);
}
}
void SDL_CloseAudio (void)
{
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
void SDL_AudioQuit(void)
{
SDL_AudioDevice *audio = current_audio;
if ( audio ) {
audio->enabled = 0;
if ( audio->thread != NULL ) {
SDL_WaitThread(audio->thread, NULL);
}
if ( audio->mixer_lock != NULL ) {
SDL_DestroyMutex(audio->mixer_lock);
}
if ( audio->fake_stream != NULL ) {
SDL_FreeAudioMem(audio->fake_stream);
}
if ( audio->convert.needed ) {
SDL_FreeAudioMem(audio->convert.buf);
}
#ifndef ENABLE_AHI
if ( audio->opened ) {
audio->CloseAudio(audio);
audio->opened = 0;
}
#endif
/* Free the driver data */
audio->free(audio);
current_audio = NULL;
}
}
#define NUM_FORMATS 6
static int format_idx;
static int format_idx_sub;
static Uint16 format_list[NUM_FORMATS][NUM_FORMATS] = {
{ AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB },
{ AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB },
{ AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U8, AUDIO_S8 },
{ AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U8, AUDIO_S8 },
};
Uint16 SDL_FirstAudioFormat(Uint16 format)
{
for ( format_idx=0; format_idx < NUM_FORMATS; ++format_idx ) {
if ( format_list[format_idx][0] == format ) {
break;
}
}
format_idx_sub = 0;
return(SDL_NextAudioFormat());
}
Uint16 SDL_NextAudioFormat(void)
{
if ( (format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS) ) {
return(0);
}
return(format_list[format_idx][format_idx_sub++]);
}
void SDL_CalculateAudioSpec(SDL_AudioSpec *spec)
{
switch (spec->format) {
case AUDIO_U8:
spec->silence = 0x80;
break;
default:
spec->silence = 0x00;
break;
}
spec->size = (spec->format&0xFF)/8;
spec->size *= spec->channels;
spec->size *= spec->samples;
}