path: root/apps/codecs/adx.c

/***************************************************************************
 *             __________               __   ___.
 *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
 *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
 *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
 *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
 *                     \/            \/     \/    \/            \/
 *
 * Copyright (C) 2006-2008 Adam Gashlin (hcs)
 * Copyright (C) 2006 Jens Arnold
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
 * as published by the Free Software Foundation; either version 2
 * of the License, or (at your option) any later version.
 *
 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
 * KIND, either express or implied.
 *
 ****************************************************************************/

#include <limits.h>
#include "codeclib.h"
#include "inttypes.h"
#include "math.h"
#include "lib/fixedpoint.h"

CODEC_HEADER

/* Maximum number of bytes to process in one iteration */
#define WAV_CHUNK_SIZE (1024*2)

/* Number of times to loop looped tracks when repeat is disabled */
#define LOOP_TIMES 2

/* Length of fade-out for looped tracks (milliseconds) */
#define FADE_LENGTH 10000L

/* Default high pass filter cutoff frequency is 500 Hz.
 * Others can be set, but the default is nearly always used,
 * and there is no way to determine if another was used, anyway.
 */
static const long cutoff = 500;

static int16_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;

/* this is the codec entry point */
enum codec_status codec_main(void)
{
    int channels;
    int sampleswritten, i;
    uint8_t *buf;
    int32_t ch1_1, ch1_2, ch2_1, ch2_2; /* ADPCM history */
    size_t n;
    int endofstream; /* end of stream flag */
    uint32_t avgbytespersec;
    int looping; /* looping flag */
    int loop_count; /* number of loops done so far */
    int fade_count; /*  countdown for fadeout */
    int fade_frames; /* length of fade in frames */
    off_t start_adr, end_adr; /* loop points */
    off_t chanstart, bufoff;
    /*long coef1=0x7298L,coef2=-0x3350L;*/
    long coef1, coef2;

    /* Generic codec initialisation */
    /* we only render 16 bits */
    ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
  
next_track:
    DEBUGF("ADX: next_track\n");
    if (codec_init()) {
        return CODEC_ERROR;
    }
    DEBUGF("ADX: after init\n");
    
    /* init history */
    ch1_1=ch1_2=ch2_1=ch2_2=0;

    /* wait for track info to load */
    if (codec_wait_taginfo() != 0)
        goto request_next_track;

    codec_set_replaygain(ci->id3);
        
    /* Get header */
    DEBUGF("ADX: request initial buffer\n");
    ci->seek_buffer(0);
    buf = ci->request_buffer(&n, 0x38);
    if (!buf || n < 0x38) {
        return CODEC_ERROR;
    }
    bufoff = 0;
    DEBUGF("ADX: read size = %lx\n",(unsigned long)n);

    /* Get file header for starting offset, channel count */
    
    chanstart = ((buf[2] << 8) | buf[3]) + 4;
    channels = buf[7];
    
    /* useful for seeking and reporting current playback position */
    avgbytespersec = ci->id3->frequency * 18 * channels / 32;
    DEBUGF("avgbytespersec=%ld\n",(unsigned long)avgbytespersec);

    /* calculate filter coefficients */

    /**
     * A simple table of these coefficients would be nice, but
     * some very odd frequencies are used and if I'm going to
     * interpolate I might as well just go all the way and
     * calclate them precisely.
     * Speed is not an issue as this only needs to be done once per file.
     */
    {
        const int64_t big28 = 0x10000000LL;
        const int64_t big32 = 0x100000000LL;
        int64_t frequency = ci->id3->frequency;
        int64_t phasemultiple = cutoff*big32/frequency;

        long z;
        int64_t a;
        const int64_t b = (M_SQRT2*big28)-big28;
        int64_t c;
        int64_t d;
        
        fp_sincos((unsigned long)phasemultiple,&z);

        a = (M_SQRT2*big28)-(z*big28/LONG_MAX);

        /**
         * In the long passed to fsqrt there are only 4 nonfractional bits,
         * which is sufficient here, but this is the only reason why I don't
         * use 32 fractional bits everywhere.
         */
        d = fp_sqrt((a+b)*(a-b)/big28,28);
        c = (a-d)*big28/b;

        coef1 = (c*8192) >> 28;
        coef2 = (c*c/big28*-4096) >> 28;
        DEBUGF("ADX: samprate=%ld ",(long)frequency);
        DEBUGF("coef1 %04x ",(unsigned int)(coef1*4));
        DEBUGF("coef2 %04x\n",(unsigned int)(coef2*-4));
    }

    /* Get loop data */
    
    looping = 0; start_adr = 0; end_adr = 0;
    if (!memcmp(buf+0x10,"\x01\xF4\x03\x00",4)) {
        /* Soul Calibur 2 style (type 03) */
        DEBUGF("ADX: type 03 found\n");
        /* check if header is too small for loop data */
        if (chanstart-6 < 0x2c) looping=0;
        else {
            looping = (buf[0x18]) ||
                      (buf[0x19]) ||
                      (buf[0x1a]) ||
                      (buf[0x1b]);
            end_adr = (buf[0x28]<<24) |
                      (buf[0x29]<<16) |
                      (buf[0x2a]<<8) |
                      (buf[0x2b]);

            start_adr = (
              (buf[0x1c]<<24) |
              (buf[0x1d]<<16) |
              (buf[0x1e]<<8) |
              (buf[0x1f])
              )/32*channels*18+chanstart;
        }
    } else if (!memcmp(buf+0x10,"\x01\xF4\x04\x00",4)) {
        /* Standard (type 04) */
        DEBUGF("ADX: type 04 found\n");
        /* check if header is too small for loop data */
        if (chanstart-6 < 0x38) looping=0;
        else {
            looping = (buf[0x24]) ||
                      (buf[0x25]) ||
                      (buf[0x26]) ||
                      (buf[0x27]);
            end_adr = (buf[0x34]<<24) |
                      (buf[0x35]<<16) |
                      (buf[0x36]<<8) |
                      buf[0x37];
            start_adr = (
              (buf[0x28]<<24) |
              (buf[0x29]<<16) |
              (buf[0x2a]<<8) |
              (buf[0x2b])
              )/32*channels*18+chanstart;
        }
    } else {
        DEBUGF("ADX: error, couldn't determine ADX type\n");
        return CODEC_ERROR;
    }

    if (looping) {
        DEBUGF("ADX: looped, start: %lx end: %lx\n",start_adr,end_adr);
    } else {
        DEBUGF("ADX: not looped\n");
    }
    
    /* advance to first frame */
    DEBUGF("ADX: first frame at %lx\n",chanstart);
    bufoff = chanstart;

    /* get in position */
    ci->seek_buffer(bufoff);


    /* setup pcm buffer format */
    ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
    if (channels == 2) {
        ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
    } else if (channels == 1) {
        ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
    } else {
        DEBUGF("ADX CODEC_ERROR: more than 2 channels\n");
        return CODEC_ERROR;
    }    

    endofstream = 0;
    loop_count = 0;
    fade_count = -1; /* disable fade */
    fade_frames = 1;

    /* The main decoder loop */
        
    while (!endofstream) {
        ci->yield();
        if (ci->stop_codec || ci->new_track) {
            break;
        }
        
        /* do we need to loop? */
        if (bufoff > end_adr-18*channels && looping) {
            DEBUGF("ADX: loop!\n");
            /* check for endless looping */
            if (ci->global_settings->repeat_mode==REPEAT_ONE) {
                loop_count=0;
                fade_count = -1; /* disable fade */
            } else {
                /* otherwise start fade after LOOP_TIMES loops */
                loop_count++;
                if (loop_count >= LOOP_TIMES && fade_count < 0) {
                    /* frames to fade over */
                    fade_frames = FADE_LENGTH*ci->id3->frequency/32/1000;
                    /* volume relative to fade_frames */
                    fade_count = fade_frames;
                    DEBUGF("ADX: fade_frames = %d\n",fade_frames);
                }
            }
            bufoff = start_adr;
            ci->seek_buffer(bufoff);
        }

        /* do we need to seek? */
        if (ci->seek_time) {
            uint32_t newpos;
            
            DEBUGF("ADX: seek to %ldms\n",ci->seek_time);

            endofstream = 0;
            loop_count = 0;
            fade_count = -1; /* disable fade */
            fade_frames = 1;

            newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
                      / (1000LL*18*channels))*(18*channels);
            bufoff = chanstart + newpos;
            while (bufoff > end_adr-18*channels) {
                bufoff-=end_adr-start_adr;
                loop_count++;
            }
            ci->seek_buffer(bufoff);
            ci->seek_complete();
        }

        if (bufoff>ci->filesize-channels*18) break; /* End of stream */
        
        sampleswritten=0;
          
        while (
                /* Is there data left in the file? */
                (bufoff <= ci->filesize-(18*channels)) &&
                /* Is there space in the output buffer? */
                (sampleswritten <= WAV_CHUNK_SIZE-(32*channels)) &&
                /* Should we be looping? */
                ((!looping) || bufoff <= end_adr-18*channels))
        {
            /* decode first/only channel */
            int32_t scale;
            int32_t ch1_0, d;

            /* fetch a frame */
            buf = ci->request_buffer(&n, 18);

            if (!buf || n!=18) {
                DEBUGF("ADX: couldn't get buffer at %lx\n",
                        bufoff);
                return CODEC_ERROR;
            }

            scale = ((buf[0] << 8) | (buf[1])) +1;
  
            for (i = 2; i < 18; i++)
            {
                d = (buf[i] >> 4) & 15;
                if (d & 8) d-= 16;
                ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
                if (ch1_0 > 32767) ch1_0 = 32767;
                else if (ch1_0 < -32768) ch1_0 = -32768;
                samples[sampleswritten] = ch1_0;
                sampleswritten+=channels;
                ch1_2 = ch1_1; ch1_1 = ch1_0;

                d = buf[i] & 15;
                if (d & 8) d -= 16;
                ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
                if (ch1_0 > 32767) ch1_0 = 32767;
                else if (ch1_0 < -32768) ch1_0 = -32768; 
                samples[sampleswritten] = ch1_0;
                sampleswritten+=channels;
                ch1_2 = ch1_1; ch1_1 = ch1_0;
            }
            bufoff+=18;
            ci->advance_buffer(18);
            
            if (channels == 2) {
                /* decode second channel */
                int32_t scale;
                int32_t ch2_0, d;

                buf = ci->request_buffer(&n, 18);

                if (!buf || n!=18) {
                    DEBUGF("ADX: couldn't get buffer at %lx\n",
                            bufoff);
                    return CODEC_ERROR;
                }

                scale = ((buf[0] << 8)|(buf[1]))+1;
  
                sampleswritten-=63;

                for (i = 2; i < 18; i++)
                {
                    d = (buf[i] >> 4) & 15;
                    if (d & 8) d-= 16;
                    ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
                    if (ch2_0 > 32767) ch2_0 = 32767;
                    else if (ch2_0 < -32768) ch2_0 = -32768;
                    samples[sampleswritten] = ch2_0;
                    sampleswritten+=2;
                    ch2_2 = ch2_1; ch2_1 = ch2_0;

                    d = buf[i] & 15;
                    if (d & 8) d -= 16;
                    ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
                    if (ch2_0 > 32767) ch2_0 = 32767;
                    else if (ch2_0 < -32768) ch2_0 = -32768; 
                    samples[sampleswritten] = ch2_0;
                    sampleswritten+=2;
                    ch2_2 = ch2_1; ch2_1 = ch2_0;
                }
                bufoff+=18;
                ci->advance_buffer(18);
                sampleswritten--; /* go back to first channel's next sample */
            }

            if (fade_count>0) {
                fade_count--;
                for (i=0;i<(channels==1?32:64);i++) samples[sampleswritten-i-1]=
                  ((int32_t)samples[sampleswritten-i-1])*fade_count/fade_frames;
                if (fade_count==0) {endofstream=1; break;}
            }
        }

        if (channels == 2)
            sampleswritten >>= 1; /* make samples/channel */

        ci->pcmbuf_insert(samples, NULL, sampleswritten);
            
        ci->set_elapsed(
           ((end_adr-start_adr)*loop_count + bufoff-chanstart)*
           1000LL/avgbytespersec);
    }

request_next_track:
    if (ci->request_next_track())
        goto next_track;

    return CODEC_OK;
}