Mercurial > libavcodec.hg
view adxdec.c @ 6561:2883ce013c20 libavcodec
use common aac sample rate tables
author | aurel |
---|---|
date | Wed, 02 Apr 2008 21:41:48 +0000 |
parents | a0797336b964 |
children | a4104482ceef |
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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_decode(short *out,const unsigned char *in,PREV *prev) { int scale = AV_RB16(in); int i; int s0,s1,s2,d; // printf("%x ",scale); in+=2; s1 = prev->s1; s2 = prev->s2; for(i=0;i<16;i++) { d = in[i]; // d>>=4; if (d&8) d-=16; d = ((signed char)d >> 4); s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14; s2 = s1; s1 = av_clip_int16(s0); *out++=s1; d = in[i]; //d&=15; if (d&8) d-=16; d = ((signed char)(d<<4) >> 4); s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14; s2 = s1; s1 = av_clip_int16(s0); *out++=s1; } prev->s1 = s1; prev->s2 = s2; } static void adx_decode_stereo(short *out,const unsigned char *in,PREV *prev) { short tmp[32*2]; int i; adx_decode(tmp ,in ,prev); adx_decode(tmp+32,in+18,prev+1); for(i=0;i<32;i++) { out[i*2] = tmp[i]; out[i*2+1] = tmp[i+32]; } } /* return data offset or 0 */ static int adx_decode_header(AVCodecContext *avctx,const unsigned char *buf,size_t bufsize) { int offset; if (buf[0]!=0x80) return 0; offset = (AV_RB32(buf)^0x80000000)+4; if (bufsize<offset || memcmp(buf+offset-6,"(c)CRI",6)) return 0; avctx->channels = buf[7]; avctx->sample_rate = AV_RB32(buf+8); avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32; return offset; } static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, const uint8_t *buf0, int buf_size) { ADXContext *c = avctx->priv_data; short *samples = data; const uint8_t *buf = buf0; int rest = buf_size; if (!c->header_parsed) { int hdrsize = adx_decode_header(avctx,buf,rest); if (hdrsize==0) return -1; c->header_parsed = 1; buf += hdrsize; rest -= hdrsize; } /* 18 bytes of data are expanded into 32*2 bytes of audio, so guard against buffer overflows */ if(rest/18 > *data_size/64) rest = (*data_size/64) * 18; if (c->in_temp) { int copysize = 18*avctx->channels - c->in_temp; memcpy(c->dec_temp+c->in_temp,buf,copysize); rest -= copysize; buf += copysize; if (avctx->channels==1) { adx_decode(samples,c->dec_temp,c->prev); samples += 32; } else { adx_decode_stereo(samples,c->dec_temp,c->prev); samples += 32*2; } } // if (avctx->channels==1) { while(rest>=18) { adx_decode(samples,buf,c->prev); rest-=18; buf+=18; samples+=32; } } else { while(rest>=18*2) { adx_decode_stereo(samples,buf,c->prev); rest-=18*2; buf+=18*2; samples+=32*2; } } // c->in_temp = rest; if (rest) { memcpy(c->dec_temp,buf,rest); buf+=rest; } *data_size = (uint8_t*)samples - (uint8_t*)data; // printf("%d:%d ",buf-buf0,*data_size); fflush(stdout); return buf-buf0; } AVCodec adpcm_adx_decoder = { "adpcm_adx", CODEC_TYPE_AUDIO, CODEC_ID_ADPCM_ADX, sizeof(ADXContext), NULL, NULL, NULL, adx_decode_frame, };