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https://github.com/joel16/SDL2.git
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Date: Sun, 29 Feb 2004 15:14:22 +0200
From: Martin_Storsj Subject: Dynamic loading of ALSA I recently discovered that SDL can dynamically load ESD and aRts, and made a patch which adds this same functionality to ALSA. The update for configure.in isn't too good (it should e.g. look for libasound.so in other directories than /usr/lib), because I'm not too good at shellscripting and autoconf. The reason for using dlfcn.h and dlopen instead of SDL_LoadLibrary and SDL_LoadFunction is that libasound uses versioned symbols, and it is necessary to load the correct version using dlvsym. This isn't probably any real portability issue, because ALSA is linux-only. --HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%40866
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710da3b0ba
18
configure.in
18
configure.in
@ -295,8 +295,22 @@ CheckALSA()
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AC_CHECK_LIB(asound, snd_pcm_open, have_alsa=yes)
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])
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if test x$have_alsa = xyes; then
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CFLAGS="$CFLAGS -DALSA_SUPPORT"
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SYSTEM_LIBS="$SYSTEM_LIBS -lasound"
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AC_ARG_ENABLE(alsa-shared,
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[ --enable-alsa-shared dynamically load ALSA audio support [default=yes]],
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, enable_alsa_shared=yes)
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alsa_lib=`ls /usr/lib/libasound.so.* | head -1 | sed 's/.*\/\(.*\)/\1/'`
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if test x$use_dlopen != xyes && \
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test x$enable_alsa_shared = xyes; then
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AC_MSG_ERROR([You must have dlopen() support and use the --enable-dlopen option])
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fi
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if test x$use_dlopen = xyes && \
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test x$enable_alsa_shared = xyes && test x$alsa_lib != x; then
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CFLAGS="$CFLAGS -DALSA_SUPPORT -DALSA_DYNAMIC=\$(alsa_lib)"
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AC_SUBST(alsa_lib)
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else
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CFLAGS="$CFLAGS -DALSA_SUPPORT"
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SYSTEM_LIBS="$SYSTEM_LIBS -lasound"
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fi
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AUDIO_SUBDIRS="$AUDIO_SUBDIRS alsa"
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AUDIO_DRIVERS="$AUDIO_DRIVERS alsa/libaudio_alsa.la"
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else
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@ -4,6 +4,8 @@
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noinst_LTLIBRARIES = libaudio_alsa.la
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libaudio_alsa_la_SOURCES = $(SRCS)
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alsa_lib = \"@alsa_lib@\"
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# The SDL audio driver sources
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SRCS = SDL_alsa_audio.c \
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SDL_alsa_audio.h
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@ -41,6 +41,16 @@
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#include "SDL_timer.h"
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#include "SDL_alsa_audio.h"
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#ifdef ALSA_DYNAMIC
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#define __USE_GNU
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#include <dlfcn.h>
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#include "SDL_name.h"
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#include "SDL_loadso.h"
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#else
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#define SDL_NAME(X) X
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#endif
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/* The tag name used by ALSA audio */
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#define DRIVER_NAME "alsa"
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@ -54,6 +64,99 @@ static void ALSA_PlayAudio(_THIS);
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static Uint8 *ALSA_GetAudioBuf(_THIS);
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static void ALSA_CloseAudio(_THIS);
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#ifdef ALSA_DYNAMIC
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static const char *alsa_library = ALSA_DYNAMIC;
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static void *alsa_handle = NULL;
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static int alsa_loaded = 0;
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static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
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static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
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static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);
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static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
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static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm);
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static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);
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static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);
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static const char *(*SDL_NAME(snd_strerror))(int errnum);
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static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);
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static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
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static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);
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static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
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static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
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static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params);
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static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
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static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir);
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static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
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static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
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static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);
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#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)
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static struct {
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const char *name;
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void **func;
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} alsa_functions[] = {
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{ "snd_pcm_open", (void**)&SDL_NAME(snd_pcm_open) },
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{ "snd_pcm_close", (void**)&SDL_NAME(snd_pcm_close) },
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{ "snd_pcm_writei", (void**)&SDL_NAME(snd_pcm_writei) },
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{ "snd_pcm_resume", (void**)&SDL_NAME(snd_pcm_resume) },
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{ "snd_pcm_prepare", (void**)&SDL_NAME(snd_pcm_prepare) },
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{ "snd_pcm_drain", (void**)&SDL_NAME(snd_pcm_drain) },
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{ "snd_strerror", (void**)&SDL_NAME(snd_strerror) },
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{ "snd_pcm_hw_params_sizeof", (void**)&SDL_NAME(snd_pcm_hw_params_sizeof) },
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{ "snd_pcm_hw_params_any", (void**)&SDL_NAME(snd_pcm_hw_params_any) },
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{ "snd_pcm_hw_params_set_access", (void**)&SDL_NAME(snd_pcm_hw_params_set_access) },
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{ "snd_pcm_hw_params_set_format", (void**)&SDL_NAME(snd_pcm_hw_params_set_format) },
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{ "snd_pcm_hw_params_set_channels", (void**)&SDL_NAME(snd_pcm_hw_params_set_channels) },
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{ "snd_pcm_hw_params_get_channels", (void**)&SDL_NAME(snd_pcm_hw_params_get_channels) },
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{ "snd_pcm_hw_params_set_rate_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_rate_near) },
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{ "snd_pcm_hw_params_set_period_size_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) },
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{ "snd_pcm_hw_params_set_periods_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_periods_near) },
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{ "snd_pcm_hw_params", (void**)&SDL_NAME(snd_pcm_hw_params) },
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{ "snd_pcm_nonblock", (void**)&SDL_NAME(snd_pcm_nonblock) },
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};
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static void UnloadALSALibrary(void) {
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if (alsa_loaded) {
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/* SDL_UnloadObject(alsa_handle);*/
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dlclose(alsa_handle);
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alsa_handle = NULL;
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alsa_loaded = 0;
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}
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}
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static int LoadALSALibrary(void) {
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int i, retval = -1;
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/* alsa_handle = SDL_LoadObject(alsa_library);*/
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alsa_handle = dlopen(alsa_library,RTLD_NOW);
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if (alsa_handle) {
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alsa_loaded = 1;
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retval = 0;
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for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) {
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/* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/
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*alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9");
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if (!*alsa_functions[i].func) {
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retval = -1;
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UnloadALSALibrary();
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break;
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}
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}
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}
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return retval;
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}
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#else
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static void UnloadALSALibrary(void) {
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return;
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}
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static int LoadALSALibrary(void) {
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return 0;
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}
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#endif /* ALSA_DYNAMIC */
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static const char *get_audio_device()
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{
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const char *device;
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@ -74,11 +177,15 @@ static int Audio_Available(void)
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snd_pcm_t *handle;
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available = 0;
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status = snd_pcm_open(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if (LoadALSALibrary() < 0) {
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return available;
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}
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status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if ( status >= 0 ) {
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available = 1;
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snd_pcm_close(handle);
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SDL_NAME(snd_pcm_close)(handle);
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}
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UnloadALSALibrary();
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return(available);
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}
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@ -86,6 +193,7 @@ static void Audio_DeleteDevice(SDL_AudioDevice *device)
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{
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free(device->hidden);
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free(device);
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UnloadALSALibrary();
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}
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static SDL_AudioDevice *Audio_CreateDevice(int devindex)
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@ -93,6 +201,7 @@ static SDL_AudioDevice *Audio_CreateDevice(int devindex)
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SDL_AudioDevice *this;
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/* Initialize all variables that we clean on shutdown */
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LoadALSALibrary();
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this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
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if ( this ) {
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memset(this, 0, (sizeof *this));
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@ -150,7 +259,7 @@ static void ALSA_PlayAudio(_THIS)
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sample_len = this->spec.samples;
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sample_buf = (signed short *)mixbuf;
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while ( sample_len > 0 ) {
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status = snd_pcm_writei(pcm_handle, sample_buf, sample_len);
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status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);
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if ( status < 0 ) {
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if ( status == -EAGAIN ) {
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SDL_Delay(1);
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@ -159,11 +268,11 @@ static void ALSA_PlayAudio(_THIS)
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if ( status == -ESTRPIPE ) {
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do {
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SDL_Delay(1);
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status = snd_pcm_resume(pcm_handle);
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status = SDL_NAME(snd_pcm_resume)(pcm_handle);
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} while ( status == -EAGAIN );
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}
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if ( status < 0 ) {
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status = snd_pcm_prepare(pcm_handle);
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status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
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}
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if ( status < 0 ) {
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/* Hmm, not much we can do - abort */
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@ -189,8 +298,8 @@ static void ALSA_CloseAudio(_THIS)
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mixbuf = NULL;
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}
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if ( pcm_handle ) {
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snd_pcm_drain(pcm_handle);
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snd_pcm_close(pcm_handle);
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SDL_NAME(snd_pcm_drain)(pcm_handle);
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SDL_NAME(snd_pcm_close)(pcm_handle);
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pcm_handle = NULL;
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}
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}
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@ -204,25 +313,25 @@ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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Uint16 test_format;
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/* Open the audio device */
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status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if ( status < 0 ) {
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SDL_SetError("Couldn't open audio device: %s", snd_strerror(status));
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SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
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return(-1);
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}
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/* Figure out what the hardware is capable of */
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snd_pcm_hw_params_alloca(¶ms);
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status = snd_pcm_hw_params_any(pcm_handle, params);
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status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params);
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if ( status < 0 ) {
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SDL_SetError("Couldn't get hardware config: %s", snd_strerror(status));
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SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
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return(-1);
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}
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/* SDL only uses interleaved sample output */
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status = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
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if ( status < 0 ) {
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SDL_SetError("Couldn't set interleaved access: %s", snd_strerror(status));
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SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
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return(-1);
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}
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@ -255,7 +364,7 @@ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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break;
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}
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if ( format != 0 ) {
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status = snd_pcm_hw_params_set_format(pcm_handle, params, format);
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status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format);
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}
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if ( status < 0 ) {
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test_format = SDL_NextAudioFormat();
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@ -269,9 +378,9 @@ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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spec->format = test_format;
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/* Set the number of channels */
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status = snd_pcm_hw_params_set_channels(pcm_handle, params, spec->channels);
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status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels);
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if ( status < 0 ) {
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status = snd_pcm_hw_params_get_channels(params);
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status = SDL_NAME(snd_pcm_hw_params_get_channels)(params);
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if ( (status <= 0) || (status > 2) ) {
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SDL_SetError("Couldn't set audio channels");
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ALSA_CloseAudio(this);
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@ -281,9 +390,9 @@ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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}
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/* Set the audio rate */
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status = snd_pcm_hw_params_set_rate_near(pcm_handle, params, spec->freq, NULL);
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status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL);
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if ( status < 0 ) {
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SDL_SetError("Couldn't set audio frequency: %s", snd_strerror(status));
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SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
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return(-1);
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}
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@ -291,14 +400,14 @@ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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/* Set the buffer size, in samples */
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frames = spec->samples;
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frames = snd_pcm_hw_params_set_period_size_near(pcm_handle, params, frames, NULL);
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frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL);
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spec->samples = frames;
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snd_pcm_hw_params_set_periods_near(pcm_handle, params, 2, NULL);
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SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL);
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/* "set" the hardware with the desired parameters */
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status = snd_pcm_hw_params(pcm_handle, params);
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status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params);
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if ( status < 0 ) {
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SDL_SetError("Couldn't set audio parameters: %s", snd_strerror(status));
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SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status));
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ALSA_CloseAudio(this);
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return(-1);
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}
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@ -319,7 +428,7 @@ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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parent = getpid();
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/* Switch to blocking mode for playback */
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snd_pcm_nonblock(pcm_handle, 0);
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SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
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/* We're ready to rock and roll. :-) */
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return(0);
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