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-rw-r--r--src/audio/sun/SDL_sunaudio.c432
1 files changed, 432 insertions, 0 deletions
diff --git a/src/audio/sun/SDL_sunaudio.c b/src/audio/sun/SDL_sunaudio.c
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--- /dev/null
+++ b/src/audio/sun/SDL_sunaudio.c
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+/*
+ SDL - Simple DirectMedia Layer
+ Copyright (C) 1997-2009 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
+};
+
+#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 ( spec->format & 0xFF ) {
+
+ 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: {
+ 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);
+}