Mercurial > sdl-ios-xcode
view src/audio/alsa/SDL_alsa_audio.c @ 3798:c8b3d3d13ed1 SDL-ryan-multiple-audio-device
Audio bootstraps can now specify that a driver is only to be used if
explicitly requested (for things like the "disk" driver that is always
available but you would never want to default to using).
Trimmed out code that can be handled by stubs in the core. The "dummy" driver
is pretty damned small now. :)
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Wed, 04 Oct 2006 21:27:53 +0000 |
parents | adf732f1f016 |
children | 49eadd6e8962 |
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/* 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 "SDL_timer.h" #include "SDL_audio.h" #include "../SDL_audiomem.h" #include "../SDL_audio_c.h" #include "SDL_alsa_audio.h" #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC #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 SDL_AUDIO_DRIVER_ALSA_DYNAMIC static const char *alsa_library = SDL_AUDIO_DRIVER_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 size_t(*SDL_NAME(snd_pcm_sw_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 snd_pcm_sframes_t (*SDL_NAME(snd_pcm_hw_params_get_period_size)) (const snd_pcm_hw_params_t * params); 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_get_periods)) (snd_pcm_hw_params_t * params); static int (*SDL_NAME(snd_pcm_hw_params)) (snd_pcm_t * pcm, snd_pcm_hw_params_t * params); /* */ static int (*SDL_NAME(snd_pcm_sw_params_current)) (snd_pcm_t * pcm, snd_pcm_sw_params_t * swparams); static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold)) (snd_pcm_t * pcm, snd_pcm_sw_params_t * params, snd_pcm_uframes_t val); static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min)) (snd_pcm_t * pcm, snd_pcm_sw_params_t * params, snd_pcm_uframes_t val); static int (*SDL_NAME(snd_pcm_sw_params)) (snd_pcm_t * pcm, snd_pcm_sw_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) #define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_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_sw_params_sizeof", (void **) (char *) &SDL_NAME(snd_pcm_sw_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_get_period_size", (void **) (char *) &SDL_NAME(snd_pcm_hw_params_get_period_size)}, { "snd_pcm_hw_params_set_periods_near", (void **) (char *) &SDL_NAME(snd_pcm_hw_params_set_periods_near)}, { "snd_pcm_hw_params_get_periods", (void **) (char *) &SDL_NAME(snd_pcm_hw_params_get_periods)}, { "snd_pcm_hw_params", (void **) (char *) &SDL_NAME(snd_pcm_hw_params)}, { "snd_pcm_sw_params_current", (void **) (char *) &SDL_NAME(snd_pcm_sw_params_current)}, { "snd_pcm_sw_params_set_start_threshold", (void **) (char *) &SDL_NAME(snd_pcm_sw_params_set_start_threshold)}, { "snd_pcm_sw_params_set_avail_min", (void **) (char *) &SDL_NAME(snd_pcm_sw_params_set_avail_min)}, { "snd_pcm_sw_params", (void **) (char *) &SDL_NAME(snd_pcm_sw_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_arraysize(alsa_functions); i++) { /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ #if HAVE_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 /* 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; } /* 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) { SDL_free(device->hidden); SDL_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 *) SDL_malloc(sizeof(SDL_AudioDevice)); if (this) { SDL_memset(this, 0, (sizeof *this)); this->hidden = (struct SDL_PrivateAudioData *) SDL_malloc((sizeof *this->hidden)); } if ((this == NULL) || (this->hidden == NULL)) { SDL_OutOfMemory(); if (this) { SDL_free(this); } return (0); } SDL_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, 0 }; /* 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; } } } } /* * 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 *) 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->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_PlayAudio(_THIS) { int status; int sample_len; signed short *sample_buf; swizzle_alsa_channels(this); 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 *hwparams; snd_pcm_sw_params_t *swparams; snd_pcm_format_t format; snd_pcm_uframes_t frames; SDL_AudioFormat 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(&hwparams); status = SDL_NAME(snd_pcm_hw_params_any) (pcm_handle, hwparams); 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, hwparams, 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);) { 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 = SDL_NAME(snd_pcm_hw_params_set_format) (pcm_handle, hwparams, 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, hwparams, spec->channels); if (status < 0) { status = SDL_NAME(snd_pcm_hw_params_get_channels) (hwparams); 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, hwparams, 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, hwparams, frames, NULL); spec->samples = frames; SDL_NAME(snd_pcm_hw_params_set_periods_near) (pcm_handle, hwparams, 2, NULL); /* "set" the hardware with the desired parameters */ status = SDL_NAME(snd_pcm_hw_params) (pcm_handle, hwparams); if (status < 0) { SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror) (status)); ALSA_CloseAudio(this); return (-1); } /* This is useful for debugging... */ /* { snd_pcm_sframes_t bufsize; int fragments; bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams); fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams); fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments); } */ /* Set the software parameters */ snd_pcm_sw_params_alloca(&swparams); status = SDL_NAME(snd_pcm_sw_params_current) (pcm_handle, swparams); if (status < 0) { SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror) (status)); ALSA_CloseAudio(this); return (-1); } status = SDL_NAME(snd_pcm_sw_params_set_start_threshold) (pcm_handle, swparams, 0); if (status < 0) { SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror) (status)); ALSA_CloseAudio(this); return (-1); } status = SDL_NAME(snd_pcm_sw_params_set_avail_min) (pcm_handle, swparams, frames); if (status < 0) { SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror) (status)); ALSA_CloseAudio(this); return (-1); } status = SDL_NAME(snd_pcm_sw_params) (pcm_handle, swparams); if (status < 0) { SDL_SetError("Couldn't set software 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); } SDL_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); } /* vi: set ts=4 sw=4 expandtab: */