src/audio/sun/SDL_sunaudio.c

/* I'm gambling no one uses this audio backend...we'll see who emails.  :)  */
#error this code has not been updated for SDL 1.3.
#error if no one emails icculus at icculus.org and tells him that this
#error  code is needed, this audio backend will eventually be removed from SDL.

/*
  Simple DirectMedia Layer
  Copyright (C) 1997-2011 Sam Lantinga <slouken@libsdl.org>

  This software is provided 'as-is', without any express or implied
  warranty.  In no event will the authors be held liable for any damages
  arising from the use of this software.

  Permission is granted to anyone to use this software for any purpose,
  including commercial applications, and to alter it and redistribute it
  freely, subject to the following restrictions:

  1. The origin of this software must not be misrepresented; you must not
     claim that you wrote the original software. If you use this software
     in a product, an acknowledgment in the product documentation would be
     appreciated but is not required.
  2. Altered source versions must be plainly marked as such, and must not be
     misrepresented as being the original software.
  3. This notice may not be removed or altered from any source distribution.
*/
#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);
}

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