Mercurial > sdl-ios-xcode
view src/audio/alsa/SDL_alsa_audio.c @ 1199:2d6dc7de1145
From: Mike Frysinger <vapier@gentoo.org>
To: sdl@libsdl.org
Date: Sun, 11 Dec 2005 22:57:37 -0500
Subject: [SDL] exec stack in libsdl update
i posted back in September a patch to remove executable stacks:
http://www.devolution.com/pipermail/sdl/2005-September/070626.html
later in November, a similar patch was merged it seems:
http://www.libsdl.org/cgi/cvsweb.cgi/SDL12/src/hermes/mmx_main.asm
however, this lacks the additional output format checks that i posted in my
patch ... this isnt a problem if the hermes asm code is only ever used to
produce ELF objects, but if this is not true, then the additional checks in
my original patch will need to be merged
-mike
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Mon, 12 Dec 2005 09:13:12 +0000 |
parents | 05d4b93b911e |
children | 1cbaeee565b1 |
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line source
/* SDL - Simple DirectMedia Layer Copyright (C) 1997-2004 Sam Lantinga This library is free software; you can redistribute it and/or modify it under the terms of the GNU Library General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This library is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Library General Public License for more details. You should have received a copy of the GNU Library General Public License along with this library; if not, write to the Free Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Sam Lantinga slouken@libsdl.org */ /* Allow access to a raw mixing buffer */ #include <stdlib.h> #include <stdio.h> #include <string.h> #include <errno.h> #include <unistd.h> #include <fcntl.h> #include <signal.h> #include <sys/types.h> #include <sys/time.h> #include "SDL_audio.h" #include "SDL_error.h" #include "SDL_audiomem.h" #include "SDL_audio_c.h" #include "SDL_timer.h" #include "SDL_alsa_audio.h" #ifdef ALSA_DYNAMIC #ifdef USE_DLVSYM #define __USE_GNU #endif #include <dlfcn.h> #include "SDL_name.h" #include "SDL_loadso.h" #else #define SDL_NAME(X) X #endif /* The tag name used by ALSA audio */ #define DRIVER_NAME "alsa" /* The default ALSA audio driver */ #define DEFAULT_DEVICE "default" /* Audio driver functions */ static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); static void ALSA_WaitAudio(_THIS); static void ALSA_PlayAudio(_THIS); static Uint8 *ALSA_GetAudioBuf(_THIS); static void ALSA_CloseAudio(_THIS); #ifdef ALSA_DYNAMIC static const char *alsa_library = ALSA_DYNAMIC; static void *alsa_handle = NULL; static int alsa_loaded = 0; static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); static const char *(*SDL_NAME(snd_strerror))(int errnum); static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access); static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val); static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); 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); 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); 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); static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) /* cast funcs to char* first, to please GCC's strict aliasing rules. */ static struct { const char *name; void **func; } alsa_functions[] = { { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, { "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) }, { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, }; static void UnloadALSALibrary(void) { if (alsa_loaded) { /* SDL_UnloadObject(alsa_handle);*/ dlclose(alsa_handle); alsa_handle = NULL; alsa_loaded = 0; } } static int LoadALSALibrary(void) { int i, retval = -1; /* alsa_handle = SDL_LoadObject(alsa_library);*/ alsa_handle = dlopen(alsa_library,RTLD_NOW); if (alsa_handle) { alsa_loaded = 1; retval = 0; for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) { /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ #ifdef USE_DLVSYM *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); if (!*alsa_functions[i].func) #endif *alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name); if (!*alsa_functions[i].func) { retval = -1; UnloadALSALibrary(); break; } } } return retval; } #else static void UnloadALSALibrary(void) { return; } static int LoadALSALibrary(void) { return 0; } #endif /* ALSA_DYNAMIC */ static const char *get_audio_device(int channels) { const char *device; device = getenv("AUDIODEV"); /* Is there a standard variable name? */ if ( device == NULL ) { if (channels == 6) device = "surround51"; else if (channels == 4) device = "surround40"; else device = DEFAULT_DEVICE; } return device; } /* Audio driver bootstrap functions */ static int Audio_Available(void) { int available; int status; snd_pcm_t *handle; available = 0; if (LoadALSALibrary() < 0) { return available; } status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if ( status >= 0 ) { available = 1; SDL_NAME(snd_pcm_close)(handle); } UnloadALSALibrary(); return(available); } static void Audio_DeleteDevice(SDL_AudioDevice *device) { free(device->hidden); free(device); UnloadALSALibrary(); } static SDL_AudioDevice *Audio_CreateDevice(int devindex) { SDL_AudioDevice *this; /* Initialize all variables that we clean on shutdown */ LoadALSALibrary(); this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice)); if ( this ) { memset(this, 0, (sizeof *this)); this->hidden = (struct SDL_PrivateAudioData *) malloc((sizeof *this->hidden)); } if ( (this == NULL) || (this->hidden == NULL) ) { SDL_OutOfMemory(); if ( this ) { free(this); } return(0); } memset(this->hidden, 0, (sizeof *this->hidden)); /* Set the function pointers */ this->OpenAudio = ALSA_OpenAudio; this->WaitAudio = ALSA_WaitAudio; this->PlayAudio = ALSA_PlayAudio; this->GetAudioBuf = ALSA_GetAudioBuf; this->CloseAudio = ALSA_CloseAudio; this->free = Audio_DeleteDevice; return this; } AudioBootStrap ALSA_bootstrap = { DRIVER_NAME, "ALSA 0.9 PCM audio", Audio_Available, Audio_CreateDevice }; /* This function waits until it is possible to write a full sound buffer */ static void ALSA_WaitAudio(_THIS) { /* Check to see if the thread-parent process is still alive */ { static int cnt = 0; /* Note that this only works with thread implementations that use a different process id for each thread. */ if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ if ( kill(parent, 0) < 0 ) { this->enabled = 0; } } } } static void ALSA_PlayAudio(_THIS) { int status; int sample_len; signed short *sample_buf; sample_len = this->spec.samples; sample_buf = (signed short *)mixbuf; while ( sample_len > 0 ) { status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); if ( status < 0 ) { if ( status == -EAGAIN ) { SDL_Delay(1); continue; } if ( status == -ESTRPIPE ) { do { SDL_Delay(1); status = SDL_NAME(snd_pcm_resume)(pcm_handle); } while ( status == -EAGAIN ); } if ( status < 0 ) { status = SDL_NAME(snd_pcm_prepare)(pcm_handle); } if ( status < 0 ) { /* Hmm, not much we can do - abort */ this->enabled = 0; return; } continue; } sample_buf += status * this->spec.channels; sample_len -= status; } } static Uint8 *ALSA_GetAudioBuf(_THIS) { return(mixbuf); } static void ALSA_CloseAudio(_THIS) { if ( mixbuf != NULL ) { SDL_FreeAudioMem(mixbuf); mixbuf = NULL; } if ( pcm_handle ) { SDL_NAME(snd_pcm_drain)(pcm_handle); SDL_NAME(snd_pcm_close)(pcm_handle); pcm_handle = NULL; } } static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) { int status; snd_pcm_hw_params_t *params; snd_pcm_format_t format; snd_pcm_uframes_t frames; Uint16 test_format; /* Open the audio device */ /* Name of device should depend on # channels in spec */ status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if ( status < 0 ) { SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); return(-1); } /* Figure out what the hardware is capable of */ snd_pcm_hw_params_alloca(¶ms); status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params); if ( status < 0 ) { SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); ALSA_CloseAudio(this); return(-1); } /* SDL only uses interleaved sample output */ status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); if ( status < 0 ) { SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); ALSA_CloseAudio(this); return(-1); } /* Try for a closest match on audio format */ status = -1; for ( test_format = SDL_FirstAudioFormat(spec->format); test_format && (status < 0); ) { switch ( test_format ) { case AUDIO_U8: format = SND_PCM_FORMAT_U8; break; case AUDIO_S8: format = SND_PCM_FORMAT_S8; break; case AUDIO_S16LSB: format = SND_PCM_FORMAT_S16_LE; break; case AUDIO_S16MSB: format = SND_PCM_FORMAT_S16_BE; break; case AUDIO_U16LSB: format = SND_PCM_FORMAT_U16_LE; break; case AUDIO_U16MSB: format = SND_PCM_FORMAT_U16_BE; break; default: format = 0; break; } if ( format != 0 ) { status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format); } if ( status < 0 ) { test_format = SDL_NextAudioFormat(); } } if ( status < 0 ) { SDL_SetError("Couldn't find any hardware audio formats"); ALSA_CloseAudio(this); return(-1); } spec->format = test_format; /* Set the number of channels */ status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels); if ( status < 0 ) { status = SDL_NAME(snd_pcm_hw_params_get_channels)(params); if ( (status <= 0) || (status > 2) ) { SDL_SetError("Couldn't set audio channels"); ALSA_CloseAudio(this); return(-1); } spec->channels = status; } /* Set the audio rate */ status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL); if ( status < 0 ) { SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); ALSA_CloseAudio(this); return(-1); } spec->freq = status; /* Set the buffer size, in samples */ frames = spec->samples; frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL); spec->samples = frames; SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL); /* "set" the hardware with the desired parameters */ status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params); if ( status < 0 ) { SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status)); ALSA_CloseAudio(this); return(-1); } /* Calculate the final parameters for this audio specification */ SDL_CalculateAudioSpec(spec); /* Allocate mixing buffer */ mixlen = spec->size; mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); if ( mixbuf == NULL ) { ALSA_CloseAudio(this); return(-1); } memset(mixbuf, spec->silence, spec->size); /* Get the parent process id (we're the parent of the audio thread) */ parent = getpid(); /* Switch to blocking mode for playback */ SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); /* We're ready to rock and roll. :-) */ return(0); }