| /* |
| SDL - Simple DirectMedia Layer |
| Copyright (C) 1997-2011 Sam Lantinga |
| |
| This library is free software; you can redistribute it and/or |
| modify it under the terms of the GNU Lesser General Public |
| License as published by the Free Software Foundation; either |
| version 2.1 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 |
| Lesser General Public License for more details. |
| |
| You should have received a copy of the GNU Lesser General Public |
| License along with this library; if not, write to the Free Software |
| Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 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 <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" |
| |
| #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
| #include "SDL_loadso.h" |
| #endif |
| |
| /* The tag name used by ALSA audio */ |
| #define DRIVER_NAME "alsa" |
| |
| 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_recover) (snd_pcm_t *, int, int); |
| 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 void (*ALSA_snd_pcm_hw_params_copy) |
| (snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *); |
| 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 *, unsigned int *); |
| static int (*ALSA_snd_pcm_hw_params_set_rate_near) |
| (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *); |
| static int (*ALSA_snd_pcm_hw_params_set_period_size_near) |
| (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *); |
| static int (*ALSA_snd_pcm_hw_params_get_period_size) |
| (const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *); |
| static 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) |
| (const snd_pcm_hw_params_t *, unsigned int *, int *); |
| static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near) |
| (snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *); |
| static int (*ALSA_snd_pcm_hw_params_get_buffer_size) |
| (const snd_pcm_hw_params_t *, snd_pcm_uframes_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) (snd_pcm_t *, snd_pcm_sw_params_t *); |
| static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int); |
| static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int); |
| |
| |
| #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
| #define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof |
| #define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof |
| |
| 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) |
| { |
| *addr = SDL_LoadFunction(alsa_handle, fn); |
| if (*addr == NULL) { |
| /* Don't call SDL_SetError(): SDL_LoadFunction already did. */ |
| 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_recover); |
| 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_copy); |
| 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_set_buffer_size_near); |
| SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size); |
| 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); |
| SDL_ALSA_SYM(snd_pcm_nonblock); |
| SDL_ALSA_SYM(snd_pcm_wait); |
| return 0; |
| } |
| |
| #undef SDL_ALSA_SYM |
| |
| #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
| |
| static void |
| UnloadALSALibrary(void) |
| { |
| if (alsa_handle != NULL) { |
| SDL_UnloadObject(alsa_handle); |
| alsa_handle = NULL; |
| } |
| } |
| |
| static int |
| LoadALSALibrary(void) |
| { |
| int retval = 0; |
| if (alsa_handle == NULL) { |
| alsa_handle = SDL_LoadObject(alsa_library); |
| if (alsa_handle == NULL) { |
| retval = -1; |
| /* Don't call SDL_SetError(): SDL_LoadObject already did. */ |
| } 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) { |
| switch (channels) { |
| case 6: |
| device = "plug:surround51"; |
| break; |
| case 4: |
| device = "plug:surround40"; |
| break; |
| default: |
| device = "default"; |
| break; |
| } |
| } |
| return device; |
| } |
| |
| |
| /* This function waits until it is possible to write a full sound buffer */ |
| static void |
| ALSA_WaitDevice(_THIS) |
| { |
| /* We're in blocking mode, so there's nothing to do here */ |
| } |
| |
| |
| /* !!! 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; \ |
| 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->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; |
| const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf; |
| const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) * |
| this->spec.channels; |
| snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples); |
| |
| swizzle_alsa_channels(this); |
| |
| while ( frames_left > 0 && this->enabled ) { |
| /* !!! FIXME: This works, but needs more testing before going live */ |
| /*ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1);*/ |
| status = ALSA_snd_pcm_writei(this->hidden->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 = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0); |
| if (status < 0) { |
| /* Hmm, not much we can do - abort */ |
| fprintf(stderr, "ALSA write failed (unrecoverable): %s\n", |
| ALSA_snd_strerror(status)); |
| this->enabled = 0; |
| return; |
| } |
| continue; |
| } |
| sample_buf += status * frame_size; |
| frames_left -= 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_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override) |
| { |
| int status; |
| snd_pcm_uframes_t bufsize; |
| |
| /* "set" the hardware with the desired parameters */ |
| status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams); |
| if ( status < 0 ) { |
| return(-1); |
| } |
| |
| /* Get samples for the actual buffer size */ |
| status = ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize); |
| if ( status < 0 ) { |
| return(-1); |
| } |
| if ( !override && bufsize != this->spec.samples * 2 ) { |
| return(-1); |
| } |
| |
| /* !!! FIXME: Is this safe to do? */ |
| this->spec.samples = bufsize / 2; |
| |
| /* This is useful for debugging */ |
| if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) { |
| snd_pcm_uframes_t persize = 0; |
| unsigned int periods = 0; |
| |
| ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL); |
| ALSA_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, 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); |
| ALSA_snd_pcm_hw_params_copy(hwparams, params); |
| |
| if ( !override ) { |
| env = SDL_getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE"); |
| if ( env ) { |
| override = SDL_atoi(env); |
| if ( override == 0 ) { |
| return(-1); |
| } |
| } |
| } |
| |
| frames = this->spec.samples; |
| status = ALSA_snd_pcm_hw_params_set_period_size_near( |
| this->hidden->pcm_handle, hwparams, &frames, NULL); |
| if ( status < 0 ) { |
| return(-1); |
| } |
| |
| periods = 2; |
| status = ALSA_snd_pcm_hw_params_set_periods_near( |
| this->hidden->pcm_handle, hwparams, &periods, NULL); |
| if ( status < 0 ) { |
| return(-1); |
| } |
| |
| return ALSA_finalize_hardware(this, hwparams, override); |
| } |
| |
| static int |
| ALSA_set_buffer_size(_THIS, 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); |
| ALSA_snd_pcm_hw_params_copy(hwparams, params); |
| |
| if ( !override ) { |
| env = SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE"); |
| if ( env ) { |
| override = SDL_atoi(env); |
| if ( override == 0 ) { |
| return(-1); |
| } |
| } |
| } |
| |
| frames = this->spec.samples * 2; |
| status = ALSA_snd_pcm_hw_params_set_buffer_size_near( |
| this->hidden->pcm_handle, hwparams, &frames); |
| if ( status < 0 ) { |
| return(-1); |
| } |
| |
| return ALSA_finalize_hardware(this, hwparams, override); |
| } |
| |
| 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; |
| SDL_AudioFormat test_format = 0; |
| unsigned int rate = 0; |
| unsigned int channels = 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); |
| channels = this->spec.channels; |
| if (status < 0) { |
| status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels); |
| if (status < 0) { |
| ALSA_CloseDevice(this); |
| SDL_SetError("ALSA: Couldn't set audio channels"); |
| return 0; |
| } |
| this->spec.channels = channels; |
| } |
| |
| /* Set the audio rate */ |
| rate = this->spec.freq; |
| status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, |
| &rate, 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 = rate; |
| |
| /* Set the buffer size, in samples */ |
| if ( ALSA_set_period_size(this, hwparams, 0) < 0 && |
| ALSA_set_buffer_size(this, hwparams, 0) < 0 ) { |
| /* Failed to set desired buffer size, do the best you can... */ |
| if ( ALSA_set_period_size(this, hwparams, 1) < 0 ) { |
| ALSA_CloseDevice(this); |
| SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status)); |
| return(-1); |
| } |
| } |
| /* 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, 1); |
| 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(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); |
| |
| /* 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; /* this audio target is available. */ |
| } |
| |
| |
| AudioBootStrap ALSA_bootstrap = { |
| DRIVER_NAME, "ALSA PCM audio", ALSA_Init, 0 |
| }; |
| |
| /* vi: set ts=4 sw=4 expandtab: */ |
