| /* |
| 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; |
| 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(&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); ) { |
| 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, 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); |
| } |