view adxenc.c @ 5875:5a61e8e2f65c libavcodec

Remove libvorbis Vorbis decoding support. Our native decoder is complete and has no known bugs, any remaining issues will hopefully be uncovered now.
author diego
date Sun, 04 Nov 2007 12:55:32 +0000
parents b1b3dd3324ae
children 48759bfbd073
line wrap: on
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/*
 * ADX ADPCM codecs
 * Copyright (c) 2001,2003 BERO
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */
#include "avcodec.h"
#include "adx.h"

/**
 * @file adx.c
 * SEGA CRI adx codecs.
 *
 * Reference documents:
 * http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
 * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
 */

/* 18 bytes <-> 32 samples */

static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
{
    int scale;
    int i;
    int s0,s1,s2,d;
    int max=0;
    int min=0;
    int data[32];

    s1 = prev->s1;
    s2 = prev->s2;
    for(i=0;i<32;i++) {
        s0 = wav[i];
        d = ((s0<<14) - SCALE1*s1 + SCALE2*s2)/BASEVOL;
        data[i]=d;
        if (max<d) max=d;
        if (min>d) min=d;
        s2 = s1;
        s1 = s0;
    }
    prev->s1 = s1;
    prev->s2 = s2;

    /* -8..+7 */

    if (max==0 && min==0) {
        memset(adx,0,18);
        return;
    }

    if (max/7>-min/8) scale = max/7;
    else scale = -min/8;

    if (scale==0) scale=1;

    AV_WB16(adx, scale);

    for(i=0;i<16;i++) {
        adx[i+2] = ((data[i*2]/scale)<<4) | ((data[i*2+1]/scale)&0xf);
    }
}

static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
{
#if 0
    struct {
        uint32_t offset; /* 0x80000000 + sample start - 4 */
        unsigned char unknown1[3]; /* 03 12 04 */
        unsigned char channel; /* 1 or 2 */
        uint32_t freq;
        uint32_t size;
        uint32_t unknown2; /* 01 f4 03 00 */
        uint32_t unknown3; /* 00 00 00 00 */
        uint32_t unknown4; /* 00 00 00 00 */

    /* if loop
        unknown3 00 15 00 01
        unknown4 00 00 00 01
        long loop_start_sample;
        long loop_start_byte;
        long loop_end_sample;
        long loop_end_byte;
        long
    */
    } adxhdr; /* big endian */
    /* offset-6 "(c)CRI" */
#endif
    AV_WB32(buf+0x00,0x80000000|0x20);
    AV_WB32(buf+0x04,0x03120400|avctx->channels);
    AV_WB32(buf+0x08,avctx->sample_rate);
    AV_WB32(buf+0x0c,0); /* FIXME: set after */
    AV_WB32(buf+0x10,0x01040300);
    AV_WB32(buf+0x14,0x00000000);
    AV_WB32(buf+0x18,0x00000000);
    memcpy(buf+0x1c,"\0\0(c)CRI",8);
    return 0x20+4;
}

static int adx_encode_init(AVCodecContext *avctx)
{
    if (avctx->channels > 2)
        return -1; /* only stereo or mono =) */
    avctx->frame_size = 32;

    avctx->coded_frame= avcodec_alloc_frame();
    avctx->coded_frame->key_frame= 1;

//    avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;

    av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");

    return 0;
}

static int adx_encode_close(AVCodecContext *avctx)
{
    av_freep(&avctx->coded_frame);

    return 0;
}

static int adx_encode_frame(AVCodecContext *avctx,
                uint8_t *frame, int buf_size, void *data)
{
    ADXContext *c = avctx->priv_data;
    const short *samples = data;
    unsigned char *dst = frame;
    int rest = avctx->frame_size;

/*
    input data size =
    ffmpeg.c: do_audio_out()
    frame_bytes = enc->frame_size * 2 * enc->channels;
*/

//    printf("sz=%d ",buf_size); fflush(stdout);
    if (!c->header_parsed) {
        int hdrsize = adx_encode_header(avctx,dst,buf_size);
        dst+=hdrsize;
        c->header_parsed = 1;
    }

    if (avctx->channels==1) {
        while(rest>=32) {
            adx_encode(dst,samples,c->prev);
            dst+=18;
            samples+=32;
            rest-=32;
        }
    } else {
        while(rest>=32*2) {
            short tmpbuf[32*2];
            int i;

            for(i=0;i<32;i++) {
                tmpbuf[i] = samples[i*2];
                tmpbuf[i+32] = samples[i*2+1];
            }

            adx_encode(dst,tmpbuf,c->prev);
            adx_encode(dst+18,tmpbuf+32,c->prev+1);
            dst+=18*2;
            samples+=32*2;
            rest-=32*2;
        }
    }
    return dst-frame;
}

AVCodec adpcm_adx_encoder = {
    "adpcm_adx",
    CODEC_TYPE_AUDIO,
    CODEC_ID_ADPCM_ADX,
    sizeof(ADXContext),
    adx_encode_init,
    adx_encode_frame,
    adx_encode_close,
    NULL,
};