/*

 * OpenAL Bridge - a simple portable library for OpenAL interface

 * Copyright (c) 2009 Vittorio Giovara <vittorio.giovara@gmail.com>,

 *                    Mario Liebisch <mario.liebisch+hw@googlemail.com>

 *

 * This program 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; version 2 of the License

 *

 * This program 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 program; if not, write to the Free Software

 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA

 */

extern ALuint sources[MAX_SOURCES];

extern const ALfloat NV[3];

extern const ALfloat LO[6];

/*SSOUND STUFF HERE*/

char SSound_load (SSound_t* pSound, const char* cFilename) {

        uint32_t magic;

        ALenum format;

        ALsizei bitsize, freq;

        char *data;

        FILE* fp;

        snprintf(pSound->Filename, 256, "%s", cFilename);

        pSound->source = -1;

        alGenBuffers(1, &pSound->Buffer);

        if(alGetError() != AL_NO_ERROR) {

                fprintf(stderr, "CSound: Couldn't create buffer.\n");

                return 0;

        }

        fp = fopen(pSound->Filename, "rb");

        if(!fp) {

                fprintf(stderr, "CSound: Couldn't open file for reading.\n");

                return 0;

        }

        if(fread(&magic, sizeof(uint32_t), 1, fp) < 1)

        {

                fclose(fp);

                fprintf(stderr, "CSound: Couldn't read file header.\n");

                return 0;

        }

        fclose(fp);

        switch (ENDIAN_BIG_32(magic)) {

                case OGG_FILE_FORMAT:

                        load_oggvorbis (pSound->Filename, &format, &data, &bitsize, &freq);

                        break;

                case WAV_FILE_FORMAT:

                        load_wavpcm (pSound->Filename, &format, &data, &bitsize, &freq);

                        break;

                default:

                        errno = EINVAL;

                        err_ret ("(%s) ERROR - File format (%08X) not supported", prog, ENDIAN_BIG_32(magic));

                        return 0;

                        break;

        }

        alBufferData(pSound->Buffer, format, data, bitsize, freq);

        if(alGetError() != AL_NO_ERROR)

        {

                fprintf(stderr, "CSound: Couldn't write buffer data.\n");

                return 0;

        }

        free(data);

        return 1;

}

void SSound_close(SSound_t* pSound)

{

        SSound_stop(pSound);

        alDeleteBuffers(1, &pSound->Buffer);

}

void SSound_play(SSound_t* pSound, const char bLoop) {

        int i;

        if(pSound->source == -1) // need a new source

        {

                int i;

                for(i = 0; i < MAX_SOURCES; i++)

                {

                        ALint state;

                        alGetSourcei(sources[i], AL_SOURCE_STATE, &state);

                        if(state != AL_PLAYING && state != AL_PAUSED)

                        {

#ifdef DEBUG

                                printf("using source %d (state 0x%x) for buffer.\n", i, state);

#endif

                                alSourceStop(sources[pSound->source]);

                                alGetError();

                                break;

                        }

                }

                if(i == MAX_SOURCES) // no available source found; skip

                {

#ifdef DEBUG

                        printf("no source to play buffer %d!\n", i);

#endif

                        return;

                }

                pSound->source = i;

        }

        else // reuse already playing source

        {

                alSourceStop(sources[pSound->source]);

        }

        alSourcei (sources[pSound->source], AL_BUFFER, pSound->Buffer);

        alSourcef (sources[pSound->source], AL_PITCH,            1.0f);

        alSourcef (sources[pSound->source], AL_GAIN,             1.0f);

        alSourcefv(sources[pSound->source], AL_POSITION, NV          );

        alSourcefv(sources[pSound->source], AL_VELOCITY, NV          );

        alSourcei (sources[pSound->source], AL_LOOPING,  bLoop       );

        alSourcePlay(sources[pSound->source]);

        if((i = alGetError()) != AL_NO_ERROR)

        {

                fprintf(stderr, "CSound: SourcePlay error 0x%4x in source %d\n", i, pSound->source);

        }

#ifdef DEBUG

        fprintf(stderr, "play %s%s [%d]\n", pSound->Filename, bLoop ? " forever" : " once", pSound->source);

#endif

}

void SSound_pause(const SSound_t* pSound) {

        if(pSound->source == -1) // not playing

                return;

        alSourcePause(sources[pSound->source]);

#ifdef DEBUG

        fprintf(stderr, "pause %s\n", pSound->Filename);

#endif

}

void SSound_continue(const SSound_t* pSound) {

        if(pSound->source == -1) // not playing

                return;

        alSourcePlay(sources[pSound->source]);

#ifdef DEBUG

        fprintf(stderr, "pause %s\n", pSound->Filename);

#endif

}

void SSound_stop(SSound_t* pSound) {

        if(pSound->source == -1) // not playing

                return;

        alSourceStop(sources[pSound->source]);

        pSound->source = -1;

#ifdef DEBUG

        fprintf(stderr, "stop %s\n", pSound->Filename);

#endif

}

void SSound_volume(const SSound_t* pSound, const float fPercentage) {

        if(pSound->source == -1) // not playing

                return;

        alSourcef(sources[pSound->source], AL_GAIN, fPercentage);

}