diff options
Diffstat (limited to 'distrib/sdl-1.2.15/src/audio/alsa/SDL_alsa_audio.c')
| -rw-r--r-- | distrib/sdl-1.2.15/src/audio/alsa/SDL_alsa_audio.c | 619 |
1 files changed, 619 insertions, 0 deletions
diff --git a/distrib/sdl-1.2.15/src/audio/alsa/SDL_alsa_audio.c b/distrib/sdl-1.2.15/src/audio/alsa/SDL_alsa_audio.c new file mode 100644 index 0000000..f10733e --- /dev/null +++ b/distrib/sdl-1.2.15/src/audio/alsa/SDL_alsa_audio.c @@ -0,0 +1,619 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 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 "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" + +/* 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_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_recover))(snd_pcm_t *pcm, int err, int silent); +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 void (*SDL_NAME(snd_pcm_hw_params_copy))(snd_pcm_hw_params_t *dst, const snd_pcm_hw_params_t *src); +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, unsigned int *val); +static 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 int (*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 int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir); +static 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))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); +static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); +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_set_avail_min))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams, snd_pcm_uframes_t val); +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))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params); +static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); +static int (*SDL_NAME(snd_pcm_wait))(snd_pcm_t *pcm, int timeout); +#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_recover", (void**)(char*)&SDL_NAME(snd_pcm_recover) }, + { "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_copy", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_copy) }, + { "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_set_buffer_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) }, + { "snd_pcm_hw_params_get_buffer_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) }, + { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, + { "snd_pcm_sw_params_set_avail_min", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_avail_min) }, + { "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", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) }, + { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, + { "snd_pcm_wait", (void**)(char*)&SDL_NAME(snd_pcm_wait) }, +}; + +static void UnloadALSALibrary(void) { + if (alsa_loaded) { + SDL_UnloadObject(alsa_handle); + alsa_handle = NULL; + alsa_loaded = 0; + } +} + +static int LoadALSALibrary(void) { + int i, retval = -1; + + alsa_handle = SDL_LoadObject(alsa_library); + 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 (!*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 ) { + switch (channels) { + case 6: + device = "plug:surround51"; + break; + case 4: + device = "plug:surround40"; + break; + default: + device = "default"; + break; + } + } + 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 PCM audio", + Audio_Available, Audio_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void ALSA_WaitAudio(_THIS) +{ + /* We're in blocking mode, so there's nothing to do here */ +} + + +/* + * 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; \ + Uint32 i; \ + for (i = 0; i < this->spec.samples; 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; + snd_pcm_uframes_t frames_left; + const Uint8 *sample_buf = (const Uint8 *) mixbuf; + const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) * this->spec.channels; + + swizzle_alsa_channels(this); + + frames_left = ((snd_pcm_uframes_t) this->spec.samples); + + while ( frames_left > 0 && this->enabled ) { + /* This works, but needs more testing before going live */ + /*SDL_NAME(snd_pcm_wait)(pcm_handle, -1);*/ + + status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left); + if ( status < 0 ) { + if ( status == -EAGAIN ) { + /* Apparently snd_pcm_recover() doesn't handle this case - does it assume snd_pcm_wait() above? */ + SDL_Delay(1); + continue; + } + status = SDL_NAME(snd_pcm_recover)(pcm_handle, status, 0); + if ( status < 0 ) { + /* Hmm, not much we can do - abort */ + fprintf(stderr, "ALSA write failed (unrecoverable): %s\n", SDL_NAME(snd_strerror)(status)); + this->enabled = 0; + return; + } + continue; + } + sample_buf += status * frame_size; + frames_left -= 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_finalize_hardware(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *hwparams, int override) +{ + int status; + snd_pcm_uframes_t bufsize; + + /* "set" the hardware with the desired parameters */ + status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams); + if ( status < 0 ) { + return(-1); + } + + /* Get samples for the actual buffer size */ + status = SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize); + if ( status < 0 ) { + return(-1); + } + if ( !override && bufsize != spec->samples * 2 ) { + return(-1); + } + + /* FIXME: Is this safe to do? */ + spec->samples = bufsize / 2; + + /* This is useful for debugging */ + if ( getenv("SDL_AUDIO_ALSA_DEBUG") ) { + snd_pcm_uframes_t persize = 0; + unsigned int periods = 0; + + SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, NULL); + SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, NULL); + + fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize); + } + return(0); +} + +static int ALSA_set_period_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params, int override) +{ + const char *env; + int status; + snd_pcm_hw_params_t *hwparams; + snd_pcm_uframes_t frames; + unsigned int periods; + + /* Copy the hardware parameters for this setup */ + snd_pcm_hw_params_alloca(&hwparams); + SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params); + + if ( !override ) { + env = getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE"); + if ( env ) { + override = SDL_atoi(env); + if ( override == 0 ) { + return(-1); + } + } + } + + frames = spec->samples; + status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL); + if ( status < 0 ) { + return(-1); + } + + periods = 2; + status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL); + if ( status < 0 ) { + return(-1); + } + + return ALSA_finalize_hardware(this, spec, hwparams, override); +} + +static int ALSA_set_buffer_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params, int override) +{ + const char *env; + int status; + snd_pcm_hw_params_t *hwparams; + snd_pcm_uframes_t frames; + + /* Copy the hardware parameters for this setup */ + snd_pcm_hw_params_alloca(&hwparams); + SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params); + + if ( !override ) { + env = getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE"); + if ( env ) { + override = SDL_atoi(env); + if ( override == 0 ) { + return(-1); + } + } + } + + frames = spec->samples * 2; + status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames); + if ( status < 0 ) { + return(-1); + } + + return ALSA_finalize_hardware(this, spec, hwparams, override); +} + +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; + unsigned int rate; + unsigned int channels; + 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); + channels = spec->channels; + if ( status < 0 ) { + status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels); + if ( status < 0 ) { + SDL_SetError("Couldn't set audio channels"); + ALSA_CloseAudio(this); + return(-1); + } + spec->channels = channels; + } + + /* Set the audio rate */ + rate = spec->freq; + + status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL); + if ( status < 0 ) { + SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + spec->freq = rate; + + /* Set the buffer size, in samples */ + if ( ALSA_set_period_size(this, spec, hwparams, 0) < 0 && + ALSA_set_buffer_size(this, spec, hwparams, 0) < 0 ) { + /* Failed to set desired buffer size, do the best you can... */ + if ( ALSA_set_period_size(this, spec, hwparams, 1) < 0 ) { + SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + } + + /* 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_avail_min)(pcm_handle, swparams, spec->samples); + if ( status < 0 ) { + SDL_SetError("Couldn't set minimum available samples: %s", SDL_NAME(snd_strerror)(status)); + ALSA_CloseAudio(this); + return(-1); + } + status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 1); + 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)(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); + + /* Switch to blocking mode for playback */ + SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); + + /* We're ready to rock and roll. :-) */ + return(0); +} |