Mercurial > sdl-ios-xcode
view src/audio/sun/SDL_sunaudio.c @ 3798:c8b3d3d13ed1 SDL-ryan-multiple-audio-device
Audio bootstraps can now specify that a driver is only to be used if
explicitly requested (for things like the "disk" driver that is always
available but you would never want to default to using).
Trimmed out code that can be handled by stubs in the core. The "dummy" driver
is pretty damned small now. :)
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Wed, 04 Oct 2006 21:27:53 +0000 |
parents | 152dcc2f089f |
children | 332a59e39ce1 |
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/* SDL - Simple DirectMedia Layer Copyright (C) 1997-2006 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 <fcntl.h> #include <errno.h> #ifdef __NETBSD__ #include <sys/ioctl.h> #include <sys/audioio.h> #endif #ifdef __SVR4 #include <sys/audioio.h> #else #include <sys/time.h> #include <sys/types.h> #endif #include <unistd.h> #include "SDL_timer.h" #include "SDL_audio.h" #include "../SDL_audiomem.h" #include "../SDL_audio_c.h" #include "../SDL_audiodev_c.h" #include "SDL_sunaudio.h" /* Open the audio device for playback, and don't block if busy */ #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) /* Audio driver functions */ static int DSP_OpenAudio(_THIS, SDL_AudioSpec * spec); static void DSP_WaitAudio(_THIS); static void DSP_PlayAudio(_THIS); static Uint8 *DSP_GetAudioBuf(_THIS); static void DSP_CloseAudio(_THIS); static Uint8 snd2au(int sample); /* Audio driver bootstrap functions */ static int Audio_Available(void) { int fd; int available; available = 0; fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 1); if (fd >= 0) { available = 1; close(fd); } return (available); } static void Audio_DeleteDevice(SDL_AudioDevice * device) { SDL_free(device->hidden); SDL_free(device); } static SDL_AudioDevice * Audio_CreateDevice(int devindex) { SDL_AudioDevice *this; /* Initialize all variables that we clean on shutdown */ 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)); audio_fd = -1; /* Set the function pointers */ this->OpenAudio = DSP_OpenAudio; this->WaitAudio = DSP_WaitAudio; this->PlayAudio = DSP_PlayAudio; this->GetAudioBuf = DSP_GetAudioBuf; this->CloseAudio = DSP_CloseAudio; this->free = Audio_DeleteDevice; return this; } AudioBootStrap SUNAUDIO_bootstrap = { "audio", "UNIX /dev/audio interface", Audio_Available, Audio_CreateDevice, 0 }; #ifdef DEBUG_AUDIO void CheckUnderflow(_THIS) { #ifdef AUDIO_GETINFO audio_info_t info; int left; ioctl(audio_fd, AUDIO_GETINFO, &info); left = (written - info.play.samples); if (written && (left == 0)) { fprintf(stderr, "audio underflow!\n"); } #endif } #endif void DSP_WaitAudio(_THIS) { #ifdef AUDIO_GETINFO #define SLEEP_FUDGE 10 /* 10 ms scheduling fudge factor */ audio_info_t info; Sint32 left; ioctl(audio_fd, AUDIO_GETINFO, &info); left = (written - info.play.samples); if (left > fragsize) { Sint32 sleepy; sleepy = ((left - fragsize) / frequency); sleepy -= SLEEP_FUDGE; if (sleepy > 0) { SDL_Delay(sleepy); } } #else fd_set fdset; FD_ZERO(&fdset); FD_SET(audio_fd, &fdset); select(audio_fd + 1, NULL, &fdset, NULL, NULL); #endif } void DSP_PlayAudio(_THIS) { /* Write the audio data */ if (ulaw_only) { /* Assuming that this->spec.freq >= 8000 Hz */ int accum, incr, pos; Uint8 *aubuf; accum = 0; incr = this->spec.freq / 8; aubuf = ulaw_buf; switch (audio_fmt & 0xFF) { case 8: { Uint8 *sndbuf; sndbuf = mixbuf; for (pos = 0; pos < fragsize; ++pos) { *aubuf = snd2au((0x80 - *sndbuf) * 64); accum += incr; while (accum > 0) { accum -= 1000; sndbuf += 1; } aubuf += 1; } } break; case 16: { Sint16 *sndbuf; sndbuf = (Sint16 *) mixbuf; for (pos = 0; pos < fragsize; ++pos) { *aubuf = snd2au(*sndbuf / 4); accum += incr; while (accum > 0) { accum -= 1000; sndbuf += 1; } aubuf += 1; } } break; } #ifdef DEBUG_AUDIO CheckUnderflow(this); #endif if (write(audio_fd, ulaw_buf, fragsize) < 0) { /* Assume fatal error, for now */ this->enabled = 0; } written += fragsize; } else { #ifdef DEBUG_AUDIO CheckUnderflow(this); #endif if (write(audio_fd, mixbuf, this->spec.size) < 0) { /* Assume fatal error, for now */ this->enabled = 0; } written += fragsize; } } Uint8 * DSP_GetAudioBuf(_THIS) { return (mixbuf); } void DSP_CloseAudio(_THIS) { if (mixbuf != NULL) { SDL_FreeAudioMem(mixbuf); mixbuf = NULL; } if (ulaw_buf != NULL) { SDL_free(ulaw_buf); ulaw_buf = NULL; } close(audio_fd); } int DSP_OpenAudio(_THIS, SDL_AudioSpec * spec) { char audiodev[1024]; #ifdef AUDIO_SETINFO int enc; #endif int desired_freq = spec->freq; /* Initialize our freeable variables, in case we fail */ audio_fd = -1; mixbuf = NULL; ulaw_buf = NULL; /* Determine the audio parameters from the AudioSpec */ switch (SDL_AUDIO_BITSIZE(spec->format)) { case 8: { /* Unsigned 8 bit audio data */ spec->format = AUDIO_U8; #ifdef AUDIO_SETINFO enc = AUDIO_ENCODING_LINEAR8; #endif } break; case 16: { /* Signed 16 bit audio data */ spec->format = AUDIO_S16SYS; #ifdef AUDIO_SETINFO enc = AUDIO_ENCODING_LINEAR; #endif } break; default: { /* !!! FIXME: fallback to conversion on unsupported types! */ SDL_SetError("Unsupported audio format"); return (-1); } } audio_fmt = spec->format; /* Open the audio device */ audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1); if (audio_fd < 0) { SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno)); return (-1); } ulaw_only = 0; /* modern Suns do support linear audio */ #ifdef AUDIO_SETINFO for (;;) { audio_info_t info; AUDIO_INITINFO(&info); /* init all fields to "no change" */ /* Try to set the requested settings */ info.play.sample_rate = spec->freq; info.play.channels = spec->channels; info.play.precision = (enc == AUDIO_ENCODING_ULAW) ? 8 : spec->format & 0xff; info.play.encoding = enc; if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0) { /* Check to be sure we got what we wanted */ if (ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) { SDL_SetError("Error getting audio parameters: %s", strerror(errno)); return -1; } if (info.play.encoding == enc && info.play.precision == (spec->format & 0xff) && info.play.channels == spec->channels) { /* Yow! All seems to be well! */ spec->freq = info.play.sample_rate; break; } } switch (enc) { case AUDIO_ENCODING_LINEAR8: /* unsigned 8bit apparently not supported here */ enc = AUDIO_ENCODING_LINEAR; spec->format = AUDIO_S16SYS; break; /* try again */ case AUDIO_ENCODING_LINEAR: /* linear 16bit didn't work either, resort to µ-law */ enc = AUDIO_ENCODING_ULAW; spec->channels = 1; spec->freq = 8000; spec->format = AUDIO_U8; ulaw_only = 1; break; default: /* oh well... */ SDL_SetError("Error setting audio parameters: %s", strerror(errno)); return -1; } } #endif /* AUDIO_SETINFO */ written = 0; /* We can actually convert on-the-fly to U-Law */ if (ulaw_only) { spec->freq = desired_freq; fragsize = (spec->samples * 1000) / (spec->freq / 8); frequency = 8; ulaw_buf = (Uint8 *) SDL_malloc(fragsize); if (ulaw_buf == NULL) { SDL_OutOfMemory(); return (-1); } spec->channels = 1; } else { fragsize = spec->samples; frequency = spec->freq / 1000; } #ifdef DEBUG_AUDIO fprintf(stderr, "Audio device %s U-Law only\n", ulaw_only ? "is" : "is not"); fprintf(stderr, "format=0x%x chan=%d freq=%d\n", spec->format, spec->channels, spec->freq); #endif /* Update the fragment size as size in bytes */ SDL_CalculateAudioSpec(spec); /* Allocate mixing buffer */ mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size); if (mixbuf == NULL) { SDL_OutOfMemory(); return (-1); } SDL_memset(mixbuf, spec->silence, spec->size); /* We're ready to rock and roll. :-) */ return (0); } /************************************************************************/ /* This function (snd2au()) copyrighted: */ /************************************************************************/ /* Copyright 1989 by Rich Gopstein and Harris Corporation */ /* */ /* Permission to use, copy, modify, and distribute this software */ /* and its documentation for any purpose and without fee is */ /* hereby granted, provided that the above copyright notice */ /* appears in all copies and that both that copyright notice and */ /* this permission notice appear in supporting documentation, and */ /* that the name of Rich Gopstein and Harris Corporation not be */ /* used in advertising or publicity pertaining to distribution */ /* of the software without specific, written prior permission. */ /* Rich Gopstein and Harris Corporation make no representations */ /* about the suitability of this software for any purpose. It */ /* provided "as is" without express or implied warranty. */ /************************************************************************/ static Uint8 snd2au(int sample) { int mask; if (sample < 0) { sample = -sample; mask = 0x7f; } else { mask = 0xff; } if (sample < 32) { sample = 0xF0 | (15 - sample / 2); } else if (sample < 96) { sample = 0xE0 | (15 - (sample - 32) / 4); } else if (sample < 224) { sample = 0xD0 | (15 - (sample - 96) / 8); } else if (sample < 480) { sample = 0xC0 | (15 - (sample - 224) / 16); } else if (sample < 992) { sample = 0xB0 | (15 - (sample - 480) / 32); } else if (sample < 2016) { sample = 0xA0 | (15 - (sample - 992) / 64); } else if (sample < 4064) { sample = 0x90 | (15 - (sample - 2016) / 128); } else if (sample < 8160) { sample = 0x80 | (15 - (sample - 4064) / 256); } else { sample = 0x80; } return (mask & sample); } /* vi: set ts=4 sw=4 expandtab: */