Mercurial > sdl-ios-xcode
view src/audio/alsa/SDL_alsa_audio.c @ 2480:b883974445fc gsoc2008_force_feedback
Some more error reporting.
Added periodic effect.
Confirmed it works.
author | Edgar Simo <bobbens@gmail.com> |
---|---|
date | Tue, 01 Jul 2008 09:22:22 +0000 |
parents | 866052b01ee5 |
children | e1da92da346c |
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 */ #include "SDL_config.h" /* Allow access to a raw mixing buffer */ #include <sys/types.h> #include <signal.h> /* For kill() */ #include <dlfcn.h> #include <errno.h> #include <string.h> #include "SDL_timer.h" #include "SDL_audio.h" #include "../SDL_audiomem.h" #include "../SDL_audio_c.h" #include "SDL_alsa_audio.h" /* The tag name used by ALSA audio */ #define DRIVER_NAME "alsa" /* The default ALSA audio driver */ #define DEFAULT_DEVICE "default" static int (*ALSA_snd_pcm_open) (snd_pcm_t **, const char *, snd_pcm_stream_t, int); static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm); static snd_pcm_sframes_t(*ALSA_snd_pcm_writei) (snd_pcm_t *, const void *, snd_pcm_uframes_t); static int (*ALSA_snd_pcm_resume) (snd_pcm_t *); static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *); static int (*ALSA_snd_pcm_drain) (snd_pcm_t *); static const char *(*ALSA_snd_strerror) (int); static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void); static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void); static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *); static int (*ALSA_snd_pcm_hw_params_set_access) (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t); static int (*ALSA_snd_pcm_hw_params_set_format) (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t); static int (*ALSA_snd_pcm_hw_params_set_channels) (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int); static int (*ALSA_snd_pcm_hw_params_get_channels) (const snd_pcm_hw_params_t *); static unsigned int (*ALSA_snd_pcm_hw_params_set_rate_near) (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int, int *); static snd_pcm_uframes_t(*ALSA_snd_pcm_hw_params_set_period_size_near) (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t, int *); static snd_pcm_sframes_t(*ALSA_snd_pcm_hw_params_get_period_size) (const snd_pcm_hw_params_t *); static unsigned int (*ALSA_snd_pcm_hw_params_set_periods_near) (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int, int *); static int (*ALSA_snd_pcm_hw_params_get_periods) (snd_pcm_hw_params_t *); static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *); static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *, snd_pcm_sw_params_t *); static int (*ALSA_snd_pcm_sw_params_set_start_threshold) (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t); static int (*ALSA_snd_pcm_sw_params_set_avail_min) (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t); static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *); static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int); #define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof #define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; static void *alsa_handle = NULL; static int load_alsa_sym(const char *fn, void **addr) { /* * !!! FIXME: * Eventually, this will deal with fallbacks, version changes, and * missing symbols we can workaround. But for now, it doesn't. */ #if HAVE_DLVSYM *addr = dlvsym(alsa_handle, fn, "ALSA_0.9"); if (*addr == NULL) #endif { *addr = dlsym(alsa_handle, fn); if (*addr == NULL) { SDL_SetError("dlsym('%s') failed: %s", fn, strerror(errno)); return 0; } } return 1; } /* cast funcs to char* first, to please GCC's strict aliasing rules. */ #define SDL_ALSA_SYM(x) \ if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1 #else #define SDL_ALSA_SYM(x) ALSA_##x = x #endif static int load_alsa_syms(void) { SDL_ALSA_SYM(snd_pcm_open); SDL_ALSA_SYM(snd_pcm_close); SDL_ALSA_SYM(snd_pcm_writei); SDL_ALSA_SYM(snd_pcm_resume); SDL_ALSA_SYM(snd_pcm_prepare); SDL_ALSA_SYM(snd_pcm_drain); SDL_ALSA_SYM(snd_strerror); SDL_ALSA_SYM(snd_pcm_hw_params_sizeof); SDL_ALSA_SYM(snd_pcm_sw_params_sizeof); SDL_ALSA_SYM(snd_pcm_hw_params_any); SDL_ALSA_SYM(snd_pcm_hw_params_set_access); SDL_ALSA_SYM(snd_pcm_hw_params_set_format); SDL_ALSA_SYM(snd_pcm_hw_params_set_channels); SDL_ALSA_SYM(snd_pcm_hw_params_get_channels); SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near); SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near); SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size); SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near); SDL_ALSA_SYM(snd_pcm_hw_params_get_periods); SDL_ALSA_SYM(snd_pcm_hw_params); SDL_ALSA_SYM(snd_pcm_sw_params_current); SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold); SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min); SDL_ALSA_SYM(snd_pcm_sw_params); SDL_ALSA_SYM(snd_pcm_nonblock); return 0; } #undef SDL_ALSA_SYM #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC static void UnloadALSALibrary(void) { if (alsa_handle != NULL) { dlclose(alsa_handle); alsa_handle = NULL; } } static int LoadALSALibrary(void) { int retval = 0; if (alsa_handle == NULL) { alsa_handle = dlopen(alsa_library, RTLD_NOW); if (alsa_handle == NULL) { retval = -1; SDL_SetError("ALSA: dlopen('%s') failed: %s\n", alsa_library, strerror(errno)); } else { retval = load_alsa_syms(); if (retval < 0) { UnloadALSALibrary(); } } } return retval; } #else static void UnloadALSALibrary(void) { } static int LoadALSALibrary(void) { load_alsa_syms(); return 0; } #endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */ static const char * get_audio_device(int channels) { const char *device; device = SDL_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; } /* This function waits until it is possible to write a full sound buffer */ static void ALSA_WaitDevice(_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. */ /* Check every 10 loops */ if (this->hidden->parent && (((++cnt) % 10) == 0)) { if (kill(this->hidden->parent, 0) < 0) { this->enabled = 0; } } } } /* !!! FIXME: is there a channel swizzler in alsalib instead? */ /* * http://bugzilla.libsdl.org/show_bug.cgi?id=110 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE * and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR" */ #define SWIZ6(T) \ T *ptr = (T *) this->hidden->mixbuf; \ const Uint32 count = (this->spec.samples / 6); \ Uint32 i; \ for (i = 0; i < count; i++, ptr += 6) { \ T tmp; \ tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \ tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \ } static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); } static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); } static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); } static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); } #undef SWIZ6 /* * Called right before feeding this->hidden->mixbuf to the hardware. Swizzle * channels from Windows/Mac order to the format alsalib will want. */ static __inline__ void swizzle_alsa_channels(_THIS) { if (this->spec.channels == 6) { const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */ if (fmtsize == 16) swizzle_alsa_channels_6_16bit(this); else if (fmtsize == 8) swizzle_alsa_channels_6_8bit(this); else if (fmtsize == 32) swizzle_alsa_channels_6_32bit(this); else if (fmtsize == 64) swizzle_alsa_channels_6_64bit(this); } /* !!! FIXME: update this for 7.1 if needed, later. */ } static void ALSA_PlayDevice(_THIS) { int status; int sample_len; signed short *sample_buf; swizzle_alsa_channels(this); sample_len = this->spec.samples; sample_buf = (signed short *) this->hidden->mixbuf; while (sample_len > 0) { status = ALSA_snd_pcm_writei(this->hidden->pcm_handle, sample_buf, sample_len); if (status < 0) { if (status == -EAGAIN) { SDL_Delay(1); continue; } if (status == -ESTRPIPE) { do { SDL_Delay(1); status = ALSA_snd_pcm_resume(this->hidden->pcm_handle); } while (status == -EAGAIN); } if (status < 0) { status = ALSA_snd_pcm_prepare(this->hidden->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_GetDeviceBuf(_THIS) { return (this->hidden->mixbuf); } static void ALSA_CloseDevice(_THIS) { if (this->hidden != NULL) { if (this->hidden->mixbuf != NULL) { SDL_FreeAudioMem(this->hidden->mixbuf); this->hidden->mixbuf = NULL; } if (this->hidden->pcm_handle) { ALSA_snd_pcm_drain(this->hidden->pcm_handle); ALSA_snd_pcm_close(this->hidden->pcm_handle); this->hidden->pcm_handle = NULL; } SDL_free(this->hidden); this->hidden = NULL; } } static int ALSA_OpenDevice(_THIS, const char *devname, int iscapture) { int status = 0; snd_pcm_t *pcm_handle = NULL; snd_pcm_hw_params_t *hwparams = NULL; snd_pcm_sw_params_t *swparams = NULL; snd_pcm_format_t format = 0; snd_pcm_uframes_t frames = 0; SDL_AudioFormat test_format = 0; /* Initialize all variables that we clean on shutdown */ this->hidden = (struct SDL_PrivateAudioData *) SDL_malloc((sizeof *this->hidden)); if (this->hidden == NULL) { SDL_OutOfMemory(); return 0; } SDL_memset(this->hidden, 0, (sizeof *this->hidden)); /* Open the audio device */ /* Name of device should depend on # channels in spec */ status = ALSA_snd_pcm_open(&pcm_handle, get_audio_device(this->spec.channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't open audio device: %s", ALSA_snd_strerror(status)); return 0; } this->hidden->pcm_handle = pcm_handle; /* Figure out what the hardware is capable of */ snd_pcm_hw_params_alloca(&hwparams); status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't get hardware config: %s", ALSA_snd_strerror(status)); return 0; } /* SDL only uses interleaved sample output */ status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't set interleaved access: %s", ALSA_snd_strerror(status)); return 0; } /* Try for a closest match on audio format */ status = -1; for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format && (status < 0);) { status = 0; /* if we can't support a format, it'll become -1. */ 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; case AUDIO_S32LSB: format = SND_PCM_FORMAT_S32_LE; break; case AUDIO_S32MSB: format = SND_PCM_FORMAT_S32_BE; break; case AUDIO_F32LSB: format = SND_PCM_FORMAT_FLOAT_LE; break; case AUDIO_F32MSB: format = SND_PCM_FORMAT_FLOAT_BE; break; default: status = -1; break; } if (status >= 0) { status = ALSA_snd_pcm_hw_params_set_format(pcm_handle, hwparams, format); } if (status < 0) { test_format = SDL_NextAudioFormat(); } } if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't find any hardware audio formats"); return 0; } this->spec.format = test_format; /* Set the number of channels */ status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams, this->spec.channels); if (status < 0) { status = ALSA_snd_pcm_hw_params_get_channels(hwparams); if ((status <= 0) || (status > 2)) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't set audio channels"); return 0; } this->spec.channels = status; } /* Set the audio rate */ status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, this->spec.freq, NULL); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't set audio frequency: %s", ALSA_snd_strerror(status)); return 0; } this->spec.freq = status; /* Set the buffer size, in samples */ frames = this->spec.samples; frames = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, frames, NULL); this->spec.samples = frames; ALSA_snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, 2, NULL); /* "set" the hardware with the desired parameters */ status = ALSA_snd_pcm_hw_params(pcm_handle, hwparams); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status)); return 0; } #if AUDIO_DEBUG { snd_pcm_sframes_t bufsize; int fragments; bufsize = ALSA_snd_pcm_hw_params_get_period_size(hwparams); fragments = ALSA_snd_pcm_hw_params_get_periods(hwparams); fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments); } #endif /* Set the software parameters */ snd_pcm_sw_params_alloca(&swparams); status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't get software config: %s", ALSA_snd_strerror(status)); return 0; } status = ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 0); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("ALSA: Couldn't set start threshold: %s", ALSA_snd_strerror(status)); return 0; } status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, frames); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("Couldn't set avail min: %s", ALSA_snd_strerror(status)); return 0; } status = ALSA_snd_pcm_sw_params(pcm_handle, swparams); if (status < 0) { ALSA_CloseDevice(this); SDL_SetError("Couldn't set software audio parameters: %s", ALSA_snd_strerror(status)); return 0; } /* Calculate the final parameters for this audio specification */ SDL_CalculateAudioSpec(&this->spec); /* Allocate mixing buffer */ this->hidden->mixlen = this->spec.size; this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); if (this->hidden->mixbuf == NULL) { ALSA_CloseDevice(this); SDL_OutOfMemory(); return 0; } SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size); /* Get the parent process id (we're the parent of the audio thread) */ this->hidden->parent = getpid(); /* Switch to blocking mode for playback */ ALSA_snd_pcm_nonblock(pcm_handle, 0); /* We're ready to rock and roll. :-) */ return 1; } static void ALSA_Deinitialize(void) { UnloadALSALibrary(); } static int ALSA_Init(SDL_AudioDriverImpl * impl) { if (LoadALSALibrary() < 0) { return 0; } /* Set the function pointers */ impl->OpenDevice = ALSA_OpenDevice; impl->WaitDevice = ALSA_WaitDevice; impl->GetDeviceBuf = ALSA_GetDeviceBuf; impl->PlayDevice = ALSA_PlayDevice; impl->CloseDevice = ALSA_CloseDevice; impl->Deinitialize = ALSA_Deinitialize; impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */ return 1; } AudioBootStrap ALSA_bootstrap = { DRIVER_NAME, "ALSA 0.9 PCM audio", ALSA_Init, 0 }; /* vi: set ts=4 sw=4 expandtab: */