pandory500/SDL-mirror
2
0
Code Issues Pull requests Projects Releases Packages Wiki Activity Actions
SDL-mirror/src/audio/sun/SDL_sunaudio.c
Ryan C. Gordon e65ddc53f0 Date: Tue, 22 Nov 2005 14:11:06 +0100 (MET) 
From: Mattias Karlsson <betasoft@acc.umu.se>
To: sdl@libsdl.org
Subject: [SDL] [PATCH] Fix (tiny) build problems on solaris (fwd)

This small patch fixes building audio on solaris 2.9 with gcc 4.0

--HG--
extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%401187
2005-11-22 15:07:41 +00:00

439 lines
10 KiB
C
Raw Blame History

/*
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
*/
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id$";
#endif
/* Allow access to a raw mixing buffer */
#include <stdlib.h>
#include <stdio.h>
#include <fcntl.h>
#include <errno.h>
#include <string.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_endian.h"
#include "SDL_audio.h"
#include "SDL_audiomem.h"
#include "SDL_audiodev_c.h"
#include "SDL_sunaudio.h"
#include "SDL_audio_c.h"
#include "SDL_timer.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);
/* 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)
{
free(device->hidden);
free(device);
}
static SDL_AudioDevice *Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
if ( this ) {
memset(this, 0, (sizeof *this));
this->hidden = (struct SDL_PrivateAudioData *)
malloc((sizeof *this->hidden));
}
if ( (this == NULL) || (this->hidden == NULL) ) {
SDL_OutOfMemory();
if ( this ) {
free(this);
}
return(0);
}
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
}
static Uint8 snd2au(int sample);
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 ) {
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 <20>-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 *)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);
}
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);
}
Reference in a new issue View git blame Copy permalink