Mercurial > sdl-ios-xcode
view src/audio/alsa/SDL_alsa_audio.c @ 1295:c3e36ac8a94c
Date: Sun, 6 Mar 2005 17:06:20 +0100
From: Per Inge Mathisen
Subject: [SDL] Fullscreen refresh on win32
Windows has a terrible default for fullscreen 3D apps of 60mhz refresh
rate. This can be fixed by the user by going into his driver's
control panel and forcing the refresh rate higher. However, this not a
very user friendly way about it, and in any case SDL contains no code
that could figure out this that condition has afflicted the user.
So the question is, could SDL fix this for the user? It is possible
under Windows to request a higher refresh rate. The danger is of
course that if the user has an old monitor, and you request a too high
refresh rate, the monitor could be damaged. However, I believe there
might be a way around that: Check before switching what refresh rate
the user's desktop runs in, and if our fullscreen dimensions are equal
or less than those of the desktop, use the higher refresh rate of 60
and the desktop rate.
Since most users run their desktops in the same or higher resolution
something sane, this should fix this problem for most users.
Thoughts?
An alternative is to add an SDL_GL_GetAttribute(SDL_GL_REFRESH_RATE)
option so that programs can bitch at their users at their own
convenience.
- Per
author | Sam Lantinga <slouken@libsdl.org> |
---|---|
date | Mon, 30 Jan 2006 06:56:10 +0000 |
parents | 05d4b93b911e |
children | 1cbaeee565b1 |
line wrap: on
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); }