/*
    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

    Carsten Griwodz
    griff@kom.tu-darmstadt.de

    based on linux/SDL_dspaudio.c by Sam Lantinga
*/
#include "SDL_config.h"

/* Allow access to a raw mixing buffer */

#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>

#include "SDL_timer.h"
#include "SDL_audio.h"
#include "SDL_stdinc.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_paudio.h"

#define DEBUG_AUDIO 0

/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
 * I guess nobody ever uses audio... Shame over AIX header files.  */
#include <sys/machine.h>
#undef BIG_ENDIAN
#include <sys/audio.h>

/* The tag name used by paud audio */
#define PAUDIO_DRIVER_NAME         "paud"

/* Open the audio device for playback, and don't block if busy */
/* #define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK) */
#define OPEN_FLAGS	O_WRONLY

/* Get the name of the audio device we use for output */

#ifndef _PATH_DEV_DSP
#define _PATH_DEV_DSP	"/dev/%caud%c/%c"
#endif

static char devsettings[][3] = {
    {'p', '0', '1'}, {'p', '0', '2'}, {'p', '0', '3'}, {'p', '0', '4'},
    {'p', '1', '1'}, {'p', '1', '2'}, {'p', '1', '3'}, {'p', '1', '4'},
    {'p', '2', '1'}, {'p', '2', '2'}, {'p', '2', '3'}, {'p', '2', '4'},
    {'p', '3', '1'}, {'p', '3', '2'}, {'p', '3', '3'}, {'p', '3', '4'},
    {'b', '0', '1'}, {'b', '0', '2'}, {'b', '0', '3'}, {'b', '0', '4'},
    {'b', '1', '1'}, {'b', '1', '2'}, {'b', '1', '3'}, {'b', '1', '4'},
    {'b', '2', '1'}, {'b', '2', '2'}, {'b', '2', '3'}, {'b', '2', '4'},
    {'b', '3', '1'}, {'b', '3', '2'}, {'b', '3', '3'}, {'b', '3', '4'},
    {'\0', '\0', '\0'}
};

static int
OpenUserDefinedDevice(char *path, int maxlen, int flags)
{
    const char *audiodev;
    int fd;

    /* Figure out what our audio device is */
    if ((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) {
        audiodev = SDL_getenv("AUDIODEV");
    }
    if (audiodev == NULL) {
        return -1;
    }
    fd = open(audiodev, flags, 0);
    if (path != NULL) {
        SDL_strlcpy(path, audiodev, maxlen);
        path[maxlen - 1] = '\0';
    }
    return fd;
}

static int
OpenAudioPath(char *path, int maxlen, int flags, int classic)
{
    struct stat sb;
    int cycle = 0;
    int fd = OpenUserDefinedDevice(path, maxlen, flags);

    if (fd != -1) {
        return fd;
    }

    /* !!! FIXME: do we really need a table here? */
    while (devsettings[cycle][0] != '\0') {
        char audiopath[1024];
        SDL_snprintf(audiopath, SDL_arraysize(audiopath),
                     _PATH_DEV_DSP,
                     devsettings[cycle][0],
                     devsettings[cycle][1], devsettings[cycle][2]);

        if (stat(audiopath, &sb) == 0) {
            fd = open(audiopath, flags, 0);
            if (fd > 0) {
                if (path != NULL) {
                    SDL_strlcpy(path, audiopath, maxlen);
                }
                return fd;
            }
        }
    }
    return -1;
}

/* This function waits until it is possible to write a full sound buffer */
static void
PAUDIO_WaitDevice(_THIS)
{
    fd_set fdset;

    /* See if we need to use timed audio synchronization */
    if (this->hidden->frame_ticks) {
        /* Use timer for general audio synchronization */
        Sint32 ticks;

        ticks = ((Sint32)(this->hidden->next_frame-SDL_GetTicks()))-FUDGE_TICKS;
        if (ticks > 0) {
            SDL_Delay(ticks);
        }
    } else {
        audio_buffer paud_bufinfo;

        /* Use select() for audio synchronization */
        struct timeval timeout;
        FD_ZERO(&fdset);
        FD_SET(this->hidden->audio_fd, &fdset);

        if (ioctl(this->hidden->audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
#ifdef DEBUG_AUDIO
            fprintf(stderr, "Couldn't get audio buffer information\n");
#endif
            timeout.tv_sec = 10;
            timeout.tv_usec = 0;
        } else {
            long ms_in_buf = paud_bufinfo.write_buf_time;
            timeout.tv_sec = ms_in_buf / 1000;
            ms_in_buf = ms_in_buf - timeout.tv_sec * 1000;
            timeout.tv_usec = ms_in_buf * 1000;
#ifdef DEBUG_AUDIO
            fprintf(stderr,
                    "Waiting for write_buf_time=%ld,%ld\n",
                    timeout.tv_sec, timeout.tv_usec);
#endif
        }

#ifdef DEBUG_AUDIO
        fprintf(stderr, "Waiting for audio to get ready\n");
#endif
        if (select(this->hidden->audio_fd+1,NULL,&fdset,NULL,&timeout) <= 0) {
            const char *message =
                "Audio timeout - buggy audio driver? (disabled)";
            /*
             * In general we should never print to the screen,
             * but in this case we have no other way of letting
             * the user know what happened.
             */
            fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
            this->enabled = 0;
            /* Don't try to close - may hang */
            this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
            fprintf(stderr, "Done disabling audio\n");
#endif
        }
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Ready!\n");
#endif
    }
}

static void
PAUDIO_PlayDevice(_THIS)
{
    int written = 0;
    const Uint8 *mixbuf = this->hidden->mixbuf;
    const size_t mixlen = this->hidden->mixlen;

    /* Write the audio data, checking for EAGAIN on broken audio drivers */
    do {
        written = write(this->hidden->audio_fd, mixbuf, mixlen);
        if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
            SDL_Delay(1);       /* Let a little CPU time go by */
        }
    }
    while ((written < 0) &&
           ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));

    /* If timer synchronization is enabled, set the next write frame */
    if (this->hidden->frame_ticks) {
        this->hidden->next_frame += this->hidden->frame_ticks;
    }

    /* If we couldn't write, assume fatal error for now */
    if (written < 0) {
        this->enabled = 0;
    }
#ifdef DEBUG_AUDIO
    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}

static Uint8 *
PAUDIO_GetDeviceBuf(_THIS)
{
    return this->hidden->mixbuf;
}

static void
PAUDIO_CloseDevice(_THIS)
{
    if (this->hidden != NULL) {
        if (this->hidden->mixbuf != NULL) {
            SDL_FreeAudioMem(this->hidden->mixbuf);
            this->hidden->mixbuf = NULL;
        }
        if (this->hidden->audio_fd >= 0) {
            close(this->hidden->audio_fd);
            this->hidden->audio_fd = -1;
        }
        SDL_free(this->hidden);
        this->hidden = NULL;
    }
}

static int
PAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
    const char *workaround = SDL_getenv("SDL_DSP_NOSELECT");
    char audiodev[1024];
    const char *err = NULL;
    int format;
    int bytes_per_sample;
    SDL_AudioFormat test_format;
    audio_init paud_init;
    audio_buffer paud_bufinfo;
    audio_status paud_status;
    audio_control paud_control;
    audio_change paud_change;
    int fd = -1;

    /* Initialize all variables that we clean on shutdown */
    this->hidden = (struct SDL_PrivateAudioData *)
                        SDL_malloc((sizeof *this->hidden));
    if (this->hidden == NULL) {
        SDL_OutOfMemory();
        return 0;
    }
    SDL_memset(this->hidden, 0, (sizeof *this->hidden));

    /* Open the audio device */
    fd = OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
    this->hidden->audio_fd = fd;
    if (fd < 0) {
        PAUDIO_CloseDevice(this);
        SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
        return 0;
    }

    /*
     * We can't set the buffer size - just ask the device for the maximum
     * that we can have.
     */
    if (ioctl(fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
        PAUDIO_CloseDevice(this);
        SDL_SetError("Couldn't get audio buffer information");
        return 0;
    }

    if (this->spec.channels > 1)
        this->spec.channels = 2;
    else
        this->spec.channels = 1;

    /*
     * Fields in the audio_init structure:
     *
     * Ignored by us:
     *
     * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
     * paud.slot_number;         * slot number of the adapter
     * paud.device_id;           * adapter identification number
     *
     * Input:
     *
     * paud.srate;           * the sampling rate in Hz
     * paud.bits_per_sample; * 8, 16, 32, ...
     * paud.bsize;           * block size for this rate
     * paud.mode;            * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
     * paud.channels;        * 1=mono, 2=stereo
     * paud.flags;           * FIXED - fixed length data
     *                       * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
     *                       * TWOS_COMPLEMENT - 2's complement data
     *                       * SIGNED - signed? comment seems wrong in sys/audio.h
     *                       * BIG_ENDIAN
     * paud.operation;       * PLAY, RECORD
     *
     * Output:
     *
     * paud.flags;           * PITCH            - pitch is supported
     *                       * INPUT            - input is supported
     *                       * OUTPUT           - output is supported
     *                       * MONITOR          - monitor is supported
     *                       * VOLUME           - volume is supported
     *                       * VOLUME_DELAY     - volume delay is supported
     *                       * BALANCE          - balance is supported
     *                       * BALANCE_DELAY    - balance delay is supported
     *                       * TREBLE           - treble control is supported
     *                       * BASS             - bass control is supported
     *                       * BESTFIT_PROVIDED - best fit returned
     *                       * LOAD_CODE        - DSP load needed
     * paud.rc;              * NO_PLAY         - DSP code can't do play requests
     *                       * NO_RECORD       - DSP code can't do record requests
     *                       * INVALID_REQUEST - request was invalid
     *                       * CONFLICT        - conflict with open's flags
     *                       * OVERLOADED      - out of DSP MIPS or memory
     * paud.position_resolution; * smallest increment for position
     */

    paud_init.srate = this->spec.freq;
    paud_init.mode = PCM;
    paud_init.operation = PLAY;
    paud_init.channels = this->spec.channels;

    /* Try for a closest match on audio format */
    format = 0;
    for (test_format = SDL_FirstAudioFormat(this->spec.format);
         !format && test_format;) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
        switch (test_format) {
        case AUDIO_U8:
            bytes_per_sample = 1;
            paud_init.bits_per_sample = 8;
            paud_init.flags = TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_S8:
            bytes_per_sample = 1;
            paud_init.bits_per_sample = 8;
            paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_S16LSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_S16MSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = BIG_ENDIAN | SIGNED | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_U16LSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_U16MSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = BIG_ENDIAN | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        default:
            break;
        }
        if (!format) {
            test_format = SDL_NextAudioFormat();
        }
    }
    if (format == 0) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Couldn't find any hardware audio formats\n");
#endif
        PAUDIO_CloseDevice(this);
        SDL_SetError("Couldn't find any hardware audio formats");
        return 0;
    }
    this->spec.format = test_format;

    /*
     * We know the buffer size and the max number of subsequent writes
     *  that can be pending. If more than one can pend, allow the application
     *  to do something like double buffering between our write buffer and
     *  the device's own buffer that we are filling with write() anyway.
     *
     * We calculate this->spec.samples like this because
     *  SDL_CalculateAudioSpec() will give put paud_bufinfo.write_buf_cap
     *  (or paud_bufinfo.write_buf_cap/2) into this->spec.size in return.
     */
    if (paud_bufinfo.request_buf_cap == 1) {
        this->spec.samples = paud_bufinfo.write_buf_cap
            / bytes_per_sample / this->spec.channels;
    } else {
        this->spec.samples = paud_bufinfo.write_buf_cap
            / bytes_per_sample / this->spec.channels / 2;
    }
    paud_init.bsize = bytes_per_sample * this->spec.channels;

    SDL_CalculateAudioSpec(&this->spec);

    /*
     * The AIX paud device init can't modify the values of the audio_init
     * structure that we pass to it. So we don't need any recalculation
     * of this stuff and no reinit call as in linux dsp and dma code.
     *
     * /dev/paud supports all of the encoding formats, so we don't need
     * to do anything like reopening the device, either.
     */
    if (ioctl(fd, AUDIO_INIT, &paud_init) < 0) {
        switch (paud_init.rc) {
        case 1:
            err = "Couldn't set audio format: DSP can't do play requests";
            break;
        case 2:
            err = "Couldn't set audio format: DSP can't do record requests";
            break;
        case 4:
            err = "Couldn't set audio format: request was invalid";
            break;
        case 5:
            err = "Couldn't set audio format: conflict with open's flags";
            break;
        case 6:
            err = "Couldn't set audio format: out of DSP MIPS or memory";
            break;
        default:
            err = "Couldn't set audio format: not documented in sys/audio.h";
            break;
        }
    }

    if (err != NULL) {
        PAUDIO_CloseDevice(this);
        SDL_SetError("Paudio: %s", err);
        return 0;
    }

    /* Allocate mixing buffer */
    this->hidden->mixlen = this->spec.size;
    this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
    if (this->hidden->mixbuf == NULL) {
        PAUDIO_CloseDevice(this);
        SDL_OutOfMemory();
        return 0;
    }
    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);

    /*
     * Set some paramters: full volume, first speaker that we can find.
     * Ignore the other settings for now.
     */
    paud_change.input = AUDIO_IGNORE;   /* the new input source */
    paud_change.output = OUTPUT_1;      /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
    paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
    paud_change.volume = 0x7fffffff;    /* volume level [0-0x7fffffff] */
    paud_change.volume_delay = AUDIO_IGNORE;    /* the new volume delay */
    paud_change.balance = 0x3fffffff;   /* the new balance */
    paud_change.balance_delay = AUDIO_IGNORE;   /* the new balance delay */
    paud_change.treble = AUDIO_IGNORE;  /* the new treble state */
    paud_change.bass = AUDIO_IGNORE;    /* the new bass state */
    paud_change.pitch = AUDIO_IGNORE;   /* the new pitch state */

    paud_control.ioctl_request = AUDIO_CHANGE;
    paud_control.request_info = (char *) &paud_change;
    if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Can't change audio display settings\n");
#endif
    }

    /*
     * Tell the device to expect data. Actual start will wait for
     * the first write() call.
     */
    paud_control.ioctl_request = AUDIO_START;
    paud_control.position = 0;
    if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
        PAUDIO_CloseDevice(this);
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Can't start audio play\n");
#endif
        SDL_SetError("Can't start audio play");
        return 0;
    }

    /* Check to see if we need to use select() workaround */
    if (workaround != NULL) {
        this->hidden->frame_ticks = (float) (this->spec.samples * 1000) /
                                                this->spec.freq;
        this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
    }

    /* We're ready to rock and roll. :-) */
    return 1;
}

static int
PAUDIO_Init(SDL_AudioDriverImpl *impl)
{
    int fd = OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
    if (fd < 0) {
        SDL_SetError("PAUDIO: Couldn't open audio device");
        return 0;
    }
    close(fd);

    /* Set the function pointers */
    impl->OpenDevice = DSP_OpenDevice;
    impl->PlayDevice = DSP_PlayDevice;
    impl->PlayDevice = DSP_WaitDevice;
    impl->GetDeviceBuf = DSP_GetDeviceBuf;
    impl->CloseDevice = DSP_CloseDevice;
    impl->OnlyHasDefaultOutputDevice = 1;  /* !!! FIXME: add device enum! */

    return 1;
}

AudioBootStrap PAUDIO_bootstrap = {
    PAUDIO_DRIVER_NAME, "AIX Paudio", PAUDIO_Init, 0
};

/* vi: set ts=4 sw=4 expandtab: */