/*

    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

};

#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);

}