view ws-snd1.c @ 5305:5892b4a6380b libavcodec

AC-3 decoder, soc revision 31, Jul 14 23:53:28 2006 UTC by cloud9 Removed _ from names Removed temporary storage for the exponents Removed ctx->samples Now each transform coefficients are stored in audio block as an array of transform coefficients for each channel added ctx->delay (output of later half of previous block) added audio_block->block_output(output of this block) I am still not able to produce the output. I checked the code twice completely. I am not missing anything in parsing or in bit allocation. Yet it throws error in getting transform coefficients sometimes. Can anyone review a code of get_transform_coeffs and help me debug it further. I think the error is in do_bit_allocation routine cuz get_transform_coeffs is dependent on the bit allocation parameters table. I have checked the bit allocation algorithm thoroughly and it is as defined in the standard. Tried everything and got stuck where to go further. Please help me.
author jbr
date Sat, 14 Jul 2007 15:42:15 +0000
parents 05e932ddaaa9
children ca944f1db2b3
line wrap: on
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/*
 * Westwood SNDx codecs
 * Copyright (c) 2005 Konstantin Shishkov
 *
 * 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"

/**
 * @file ws-snd.c
 * Westwood SNDx codecs.
 *
 * Reference documents about VQA format and its audio codecs
 * can be found here:
 * http://www.multimedia.cx
 */

static const char ws_adpcm_2bit[] = { -2, -1, 0, 1};
static const char ws_adpcm_4bit[] = {
    -9, -8, -6, -5, -4, -3, -2, -1,
     0,  1,  2,  3,  4,  5,  6,  8 };

#define CLIP8(a) if(a>127)a=127;if(a<-128)a=-128;

static int ws_snd_decode_init(AVCodecContext * avctx)
{
//    WSSNDContext *c = avctx->priv_data;

    return 0;
}

static int ws_snd_decode_frame(AVCodecContext *avctx,
                void *data, int *data_size,
                uint8_t *buf, int buf_size)
{
//    WSSNDContext *c = avctx->priv_data;

    int in_size, out_size;
    int sample = 0;
    int i;
    short *samples = data;

    if (!buf_size)
        return 0;

    out_size = AV_RL16(&buf[0]);
    *data_size = out_size * 2;
    in_size = AV_RL16(&buf[2]);
    buf += 4;

    if (in_size == out_size) {
        for (i = 0; i < out_size; i++)
            *samples++ = (*buf++ - 0x80) << 8;
        return buf_size;
    }

    while (out_size > 0) {
        int code;
        uint8_t count;
        code = (*buf) >> 6;
        count = (*buf) & 0x3F;
        buf++;
        switch(code) {
        case 0: /* ADPCM 2-bit */
            for (count++; count > 0; count--) {
                code = *buf++;
                sample += ws_adpcm_2bit[code & 0x3];
                CLIP8(sample);
                *samples++ = sample << 8;
                sample += ws_adpcm_2bit[(code >> 2) & 0x3];
                CLIP8(sample);
                *samples++ = sample << 8;
                sample += ws_adpcm_2bit[(code >> 4) & 0x3];
                CLIP8(sample);
                *samples++ = sample << 8;
                sample += ws_adpcm_2bit[(code >> 6) & 0x3];
                CLIP8(sample);
                *samples++ = sample << 8;
                out_size -= 4;
            }
            break;
        case 1: /* ADPCM 4-bit */
            for (count++; count > 0; count--) {
                code = *buf++;
                sample += ws_adpcm_4bit[code & 0xF];
                CLIP8(sample);
                *samples++ = sample << 8;
                sample += ws_adpcm_4bit[code >> 4];
                CLIP8(sample);
                *samples++ = sample << 8;
                out_size -= 2;
            }
            break;
        case 2: /* no compression */
            if (count & 0x20) { /* big delta */
                char t;
                t = count;
                t <<= 3;
                sample += t >> 3;
                *samples++ = sample << 8;
                out_size--;
            } else { /* copy */
                for (count++; count > 0; count--) {
                    *samples++ = (*buf++ - 0x80) << 8;
                    out_size--;
                }
                sample = buf[-1] - 0x80;
            }
            break;
        default: /* run */
            for(count++; count > 0; count--) {
                *samples++ = sample << 8;
                out_size--;
            }
        }
    }

    return buf_size;
}

AVCodec ws_snd1_decoder = {
    "ws_snd1",
    CODEC_TYPE_AUDIO,
    CODEC_ID_WESTWOOD_SND1,
    0,
    ws_snd_decode_init,
    NULL,
    NULL,
    ws_snd_decode_frame,
};