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author | Steve Kondik <shade@chemlab.org> | 2012-11-18 15:47:18 -0800 |
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committer | Steve Kondik <shade@chemlab.org> | 2012-11-18 15:47:18 -0800 |
commit | a546c7006355a7bd1df4267ee53d0bfa2c017c8c (patch) | |
tree | 01be0bf6c0d6968e1468ec9661fd52110f9b05a7 /distrib/sdl-1.2.12/src/audio/alsa/SDL_alsa_audio.c | |
parent | baf3d7830396202df5cc47bd7bcee109c319cdb3 (diff) | |
parent | 0f809250987b64f491bd3b4b73c0f0d33036a786 (diff) | |
download | external_qemu-a546c7006355a7bd1df4267ee53d0bfa2c017c8c.zip external_qemu-a546c7006355a7bd1df4267ee53d0bfa2c017c8c.tar.gz external_qemu-a546c7006355a7bd1df4267ee53d0bfa2c017c8c.tar.bz2 |
Merge branch 'jb-mr1-release' of https://android.googlesource.com/platform/external/qemu into mr1-staging
Change-Id: I8a4a71ac65b08e6e17f26c942f67a15b85211115
Diffstat (limited to 'distrib/sdl-1.2.12/src/audio/alsa/SDL_alsa_audio.c')
-rw-r--r-- | distrib/sdl-1.2.12/src/audio/alsa/SDL_alsa_audio.c | 538 |
1 files changed, 0 insertions, 538 deletions
diff --git a/distrib/sdl-1.2.12/src/audio/alsa/SDL_alsa_audio.c b/distrib/sdl-1.2.12/src/audio/alsa/SDL_alsa_audio.c deleted file mode 100644 index a5138d1..0000000 --- a/distrib/sdl-1.2.12/src/audio/alsa/SDL_alsa_audio.c +++ /dev/null @@ -1,538 +0,0 @@ -/* - 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); -} |