Mercurial > sdl-ios-xcode
view src/audio/alsa/SDL_alsa_audio.c @ 3795:589bc3d060cd SDL-ryan-multiple-audio-device
More 1.3 audio work...moved dsp and dma drivers over to new model. Untested!
author | Ryan C. Gordon <icculus@icculus.org> |
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date | Wed, 04 Oct 2006 06:00:10 +0000 |
parents | adf732f1f016 |
children | 5f6550e5184f c8b3d3d13ed1 |
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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 }; /* 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: */