adpcm.c

tcpmp播放器的flv插件

/*
 * ADPCM codecs
 * Copyright (c) 2001-2003 The ffmpeg Project
 *
 * 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 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */
#include "avcodec.h"
#include "bitstream.h"

/**
 * @file adpcm.c
 * ADPCM codecs.
 * First version by Francois Revol ([e-mail])
 * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
 *   by Mike Melanson ([e-mail])
 * CD-ROM XA ADPCM codec by BERO
 * EA ADPCM decoder by Robin Kay ([e-mail])
 *
 * Features and limitations:
 *
 * Reference documents:
 * http://www.pcisys.net/~melanson/codecs/simpleaudio.html
 * http://www.geocities.com/SiliconValley/8682/aud3.txt
 * http://openquicktime.sourceforge.net/plugins.htm
 * XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html
 * http://www.cs.ucla.edu/~leec/mediabench/applications.html
 * SoX source code http://home.sprynet.com/~cbagwell/sox.html
 *
 * CD-ROM XA:
 * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html
 * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html
 * readstr http://www.geocities.co.jp/Playtown/2004/
 */

#define BLKSIZE 1024

#define CLAMP_TO_SHORT(value) \
if (value > 32767) \
    value = 32767; \
else if (value < -32768) \
    value = -32768; \

/* step_table[] and index_table[] are from the ADPCM reference source */
/* This is the index table: */
static const int index_table[16] = {
    -1, -1, -1, -1, 2, 4, 6, 8,
    -1, -1, -1, -1, 2, 4, 6, 8,
};

/** 
 * This is the step table. Note that many programs use slight deviations from
 * this table, but such deviations are negligible:
 */
static const int step_table[89] = {
    7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
    19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
    50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
    130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
    337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
    876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
    2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
    5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
    15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
};

/* These are for MS-ADPCM */
/* AdaptationTable[], AdaptCoeff1[], and AdaptCoeff2[] are from libsndfile */
static const int AdaptationTable[] = {
        230, 230, 230, 230, 307, 409, 512, 614,
        768, 614, 512, 409, 307, 230, 230, 230
};

static const int AdaptCoeff1[] = {
        256, 512, 0, 192, 240, 460, 392
};

static const int AdaptCoeff2[] = {
        0, -256, 0, 64, 0, -208, -232
};

/* These are for CD-ROM XA ADPCM */
static const int xa_adpcm_table[5][2] = {
   {   0,   0 },
   {  60,   0 },
   { 115, -52 },
   {  98, -55 },
   { 122, -60 }
};

static const int ea_adpcm_table[] = {
    0, 240, 460, 392, 0, 0, -208, -220, 0, 1,
    3, 4, 7, 8, 10, 11, 0, -1, -3, -4
};

static const int ct_adpcm_table[8] = {
    0x00E6, 0x00E6, 0x00E6, 0x00E6,
    0x0133, 0x0199, 0x0200, 0x0266
};

// padded to zero where table size is less then 16
static const int swf_index_tables[4][16] = {
    /*2*/ { -1, 2 },
    /*3*/ { -1, -1, 2, 4 },
    /*4*/ { -1, -1, -1, -1, 2, 4, 6, 8 },
    /*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 }
};

static const int yamaha_indexscale[] = {
    230, 230, 230, 230, 307, 409, 512, 614,
    230, 230, 230, 230, 307, 409, 512, 614
};

static const int yamaha_difflookup[] = {
    1, 3, 5, 7, 9, 11, 13, 15,
    -1, -3, -5, -7, -9, -11, -13, -15
};

/* end of tables */

typedef struct ADPCMChannelStatus {
    int predictor;
    short int step_index;
    int step;
    /* for encoding */
    int prev_sample;

    /* MS version */
    short sample1;
    short sample2;
    int coeff1;
    int coeff2;
    int idelta;
} ADPCMChannelStatus;

typedef struct ADPCMContext {
    int channel; /* for stereo MOVs, decode left, then decode right, then tell it's decoded */
    ADPCMChannelStatus status[2];
    short sample_buffer[32]; /* hold left samples while waiting for right samples */

    /* SWF only */
    int nb_bits;
    int nb_samples;
} ADPCMContext;

/* XXX: implement encoding */

#ifdef CONFIG_ENCODERS
static int adpcm_encode_init(AVCodecContext *avctx)
{
    if (avctx->channels > 2)
        return -1; /* only stereo or mono =) */
    switch(avctx->codec->id) {
    case CODEC_ID_ADPCM_IMA_QT:
        av_log(avctx, AV_LOG_ERROR, "ADPCM: codec adpcm_ima_qt unsupported for encoding !\n");
        avctx->frame_size = 64; /* XXX: can multiple of avctx->channels * 64 (left and right blocks are interleaved) */
        return -1;
        break;
    case CODEC_ID_ADPCM_IMA_WAV:
        avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 / (4 * avctx->channels) + 1; /* each 16 bits sample gives one nibble */
                                                             /* and we have 4 bytes per channel overhead */
        avctx->block_align = BLKSIZE;
        /* seems frame_size isn't taken into account... have to buffer the samples :-( */
        break;
    case CODEC_ID_ADPCM_MS:
        avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */
                                                             /* and we have 7 bytes per channel overhead */
        avctx->block_align = BLKSIZE;
        break;
    case CODEC_ID_ADPCM_YAMAHA:
        avctx->frame_size = BLKSIZE * avctx->channels;
        avctx->block_align = BLKSIZE;
        break;
    default:
        return -1;
        break;
    }

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

    return 0;
}

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

    return 0;
}


static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, short sample)
{
    int step_index;
    unsigned char nibble;
    
    int sign = 0; /* sign bit of the nibble (MSB) */
    int delta, predicted_delta;

    delta = sample - c->prev_sample;

    if (delta < 0) {
        sign = 1;
        delta = -delta;
    }

    step_index = c->step_index;

    /* nibble = 4 * delta / step_table[step_index]; */
    nibble = (delta << 2) / step_table[step_index];

    if (nibble > 7)
        nibble = 7;

    step_index += index_table[nibble];
    if (step_index < 0)
        step_index = 0;
    if (step_index > 88)
        step_index = 88;

    /* what the decoder will find */
    predicted_delta = ((step_table[step_index] * nibble) / 4) + (step_table[step_index] / 8);

    if (sign)
        c->prev_sample -= predicted_delta;
    else
        c->prev_sample += predicted_delta;

    CLAMP_TO_SHORT(c->prev_sample);


    nibble += sign << 3; /* sign * 8 */   

    /* save back */
    c->step_index = step_index;

    return nibble;
}

static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, short sample)
{
    int predictor, nibble, bias;

    predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
    
    nibble= sample - predictor;
    if(nibble>=0) bias= c->idelta/2;
    else          bias=-c->idelta/2;
        
    nibble= (nibble + bias) / c->idelta;
    nibble= clip(nibble, -8, 7)&0x0F;
    
    predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
    CLAMP_TO_SHORT(predictor);

    c->sample2 = c->sample1;
    c->sample1 = predictor;

    c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8;
    if (c->idelta < 16) c->idelta = 16;

    return nibble;
}

static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, short sample)
{
    int i1 = 0, j1;

    if(!c->step) {
        c->predictor = 0;
        c->step = 127;
    }
    j1 = sample - c->predictor;

    j1 = (j1 * 8) / c->step;
    i1 = abs(j1) / 2;
    if (i1 > 7)
        i1 = 7;
    if (j1 < 0)
        i1 += 8;

    c->predictor = c->predictor + ((c->step * yamaha_difflookup[i1]) / 8);
    CLAMP_TO_SHORT(c->predictor);
    c->step = (c->step * yamaha_indexscale[i1]) >> 8;
    c->step = clip(c->step, 127, 24567);

    return i1;
}

static int adpcm_encode_frame(AVCodecContext *avctx,
			    unsigned char *frame, int buf_size, void *data)
{
    int n, i, st;
    short *samples;
    unsigned char *dst;
    ADPCMContext *c = avctx->priv_data;

    dst = frame;
    samples = (short *)data;
    st= avctx->channels == 2;
/*    n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */

    switch(avctx->codec->id) {
    case CODEC_ID_ADPCM_IMA_QT: /* XXX: can't test until we get .mov writer */
        break;
    case CODEC_ID_ADPCM_IMA_WAV:
        n = avctx->frame_size / 8;
            c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
/*            c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */
            *dst++ = (c->status[0].prev_sample) & 0xFF; /* little endian */
            *dst++ = (c->status[0].prev_sample >> 8) & 0xFF;
            *dst++ = (unsigned char)c->status[0].step_index;
            *dst++ = 0; /* unknown */
            samples++;
            if (avctx->channels == 2) {
                c->status[1].prev_sample = (signed short)samples[1];
/*                c->status[1].step_index = 0; */
                *dst++ = (c->status[1].prev_sample) & 0xFF;
                *dst++ = (c->status[1].prev_sample >> 8) & 0xFF;
                *dst++ = (unsigned char)c->status[1].step_index;
                *dst++ = 0;
                samples++;
            }
        
            /* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */
            for (; n>0; n--) {
                *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]) & 0x0F;
                *dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4) & 0xF0;
                dst++;
                *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]) & 0x0F;
                *dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4) & 0xF0;
                dst++;
                *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]) & 0x0F;
                *dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4) & 0xF0;
                dst++;
                *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]) & 0x0F;
                *dst |= (adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4) & 0xF0;
                dst++;
                /* right channel */
                if (avctx->channels == 2) {
                    *dst = adpcm_ima_compress_sample(&c->status[1], samples[1]);
                    *dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4;
                    dst++;
                    *dst = adpcm_ima_compress_sample(&c->status[1], samples[5]);
                    *dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4;
                    dst++;
                    *dst = adpcm_ima_compress_sample(&c->status[1], samples[9]);
                    *dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
                    dst++;
                    *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
                    *dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
                    dst++;
                }
                samples += 8 * avctx->channels;
            }
        break;
    case CODEC_ID_ADPCM_MS:
        for(i=0; i<avctx->channels; i++){
            int predictor=0;

            *dst++ = predictor;
            c->status[i].coeff1 = AdaptCoeff1[predictor];