/*
 *  audiofile.c
 *
 *	Scott Wilson
 *	rswilson@ccrma.stanford.edu
 *  last updated:  June 15, 2003
 *
 *
 */

#include "audiofile.h"
#include "codec.h"
#include "vector.h"

int audiofile_fill_fileinfo(SF_INFO *sf_info, codec_fileinfo_t *info);

codec_err_t audiofile_open(const char *filename, audiofile_mode_t mode, codec_file_p fp) {
    SNDFILE* sf;
    SF_INFO sf_info;

    if (mode == SFM_WRITE) {
        // fill sfinfo data structure
        sf_info.samplerate = fp->info.srate;
	printf("srate = %d\n", sf_info.samplerate);
        sf_info.channels = fp->info.channels;
        // conveniently kept from openign the original file with libsndfile
        sf_info.format = fp->info.sfformat;

	/*    if (! sf_format_check (&sf_info)) {
            codec_error("audiofile_open",filename,"bad soundfile format");
            return 1;
	    }*/
             
    }
    
    if (! (sf = sf_open(filename, mode, &sf_info))) {
      printf("mode is %d\n",mode);
      printf("SFM_WRITE is %d\n",SFM_WRITE);
        codec_error("audiofile_open",filename, sf_strerror(sf));
        return 1;
    }
    
    fp->info.filename = filename;
    if (audiofile_fill_fileinfo(&sf_info, &fp->info)) {
        /* was a bad file format */
        codec_error("audiofile_open", filename, "closing due to bad format");
        sf_close(sf);
        return 1;
    }

    fp->ptr.sf_p = sf;
    return ERR_NONE;
}

codec_err_t audiofile_close(codec_file_p fp) {
    int err;

    if (err = sf_close(fp->ptr.sf_p)) {
        codec_error("audiofile_close", fp->info.filename, sf_strerror(fp->ptr.sf_p));
        return err;
    }
    return ERR_NONE;
}

// vector should be a channels*blocksize nchannel vector (array of ptrs to vectors of codec_float_t)
codec_long_t audiofile_read(codec_file_p fp, codec_long_t blocksize, codec_vector_p *vector) {
    int n,m;
    sf_count_t count;
    codec_err_t err;
    codec_vector_p tmpvector;
    int chans = fp->info.channels;

    err = vector_alloc(blocksize*chans, &tmpvector);
    count = sf_readf_double(fp->ptr.sf_p, tmpvector, blocksize);
 
    for (n=0; n<count; n++)
        for (m=0; m<chans; m++)
            vector[m][n] = tmpvector[n*chans + m];

    vector_dealloc(tmpvector);
    return count;
}

// vector should be a channels*blocksize nchannel vector (array of ptrs to vectors of codec_float_t)
codec_long_t audiofile_write(codec_file_p fp, codec_long_t blocksize, codec_vector_p *vector) {
    int n,m;
    sf_count_t count;
    codec_err_t err;
    codec_vector_p tmpvector;
    int chans = fp->info.channels;

    err = vector_alloc(blocksize*chans, &tmpvector);
    // interleave the samples
    for (n=0; n<blocksize; n++)
        for (m=0; m<chans; m++)
            tmpvector[n*chans + m] = vector[m][n];

    count = sf_writef_double(fp->ptr.sf_p, tmpvector, blocksize);
    if (count != blocksize) {
        codec_error("audiofile","write",sf_strerror(fp->ptr.sf_p));
        vector_dealloc(tmpvector);
        return 0;
    }
    fp->info.frames += blocksize;

    vector_dealloc(tmpvector);
    return blocksize;
}

codec_err_t audiofile_rewind(codec_file_p fp) {
    return sf_seek(fp->ptr.sf_p, 0, SEEK_SET);
}

codec_err_t audiofile_fill_fileinfo(SF_INFO *sf_info, codec_fileinfo_t *info) {

    info->srate = sf_info->samplerate;
    info->channels = sf_info->channels;
    info->sfformat = sf_info->format;
    info->frames = sf_info->frames;
    switch (sf_info->format & 0x000f) {
        case SF_FORMAT_PCM_S8:
            info->bits_per_sample = 8;
            info->f_signed = 1;
            info->f_float = 0;
            break;
        case SF_FORMAT_PCM_U8:
            info->bits_per_sample = 8;
            info->f_signed = 0;
            info->f_float = 0;
            break;
        case SF_FORMAT_PCM_16:
            info->bits_per_sample = 16;
            info->f_signed = 1;
            info->f_float = 0;
            break;
        case SF_FORMAT_PCM_24:
            info->bits_per_sample = 24;
            info->f_signed = 1;
            info->f_float = 0;
            break;
        case SF_FORMAT_PCM_32:
            info->bits_per_sample = 32;
            info->f_signed = 1;
            info->f_float = 0;
            break;
        case SF_FORMAT_FLOAT:
            info->bits_per_sample = 32;
            info->f_signed = 1;
            info->f_float = 1;
            break;
        case SF_FORMAT_DOUBLE:
            info->bits_per_sample = 64;
            info->f_signed = 1;
            info->f_float = 1;
            break;
        default:
            fprintf(stderr, "Error, unknown sound file format\n");
            return -1;
    }
    return 0;
}