Mercurial > libavcodec.hg
view adx.c @ 5770:a1a947d67948 libavcodec
Further modularize short reference list management for upcoming PAFF implementation.
patch by Jeff Downs, heydowns a borg d com
original thread:
Subject: [FFmpeg-devel] [PATCH] Implement PAFF in H.264
Date: 18/09/07 20:30
author | andoma |
---|---|
date | Thu, 04 Oct 2007 06:38:58 +0000 |
parents | cf77b8588f15 |
children | ad587a64ef52 |
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" /** * @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/ */ typedef struct { int s1,s2; } PREV; typedef struct { PREV prev[2]; int header_parsed; unsigned char dec_temp[18*2]; unsigned short enc_temp[32*2]; int in_temp; } ADXContext; //#define BASEVOL 0x11e0 #define BASEVOL 0x4000 #define SCALE1 0x7298 #define SCALE2 0x3350 /* 18 bytes <-> 32 samples */ #ifdef CONFIG_ENCODERS 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); } } #endif //CONFIG_ENCODERS 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]; } } #ifdef CONFIG_ENCODERS 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_decode_init(AVCodecContext *avctx); 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"); adx_decode_init(avctx); 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; } #endif //CONFIG_ENCODERS static int is_adx(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; return offset; } /* return data offset or 6 */ static int adx_decode_header(AVCodecContext *avctx,const unsigned char *buf,size_t bufsize) { int offset; int channels,freq,size; offset = is_adx(buf,bufsize); if (offset==0) return 0; channels = buf[7]; freq = AV_RB32(buf+8); size = AV_RB32(buf+12); // printf("freq=%d ch=%d\n",freq,channels); avctx->sample_rate = freq; avctx->channels = channels; avctx->bit_rate = freq*channels*18*8/32; // avctx->frame_size = 18*channels; return offset; } static int adx_decode_init(AVCodecContext * avctx) { ADXContext *c = avctx->priv_data; // printf("adx_decode_init\n"); fflush(stdout); c->prev[0].s1 = 0; c->prev[0].s2 = 0; c->prev[1].s1 = 0; c->prev[1].s2 = 0; c->header_parsed = 0; c->in_temp = 0; return 0; } #if 0 static void dump(unsigned char *buf,size_t len) { int i; for(i=0;i<len;i++) { if ((i&15)==0) av_log(NULL, AV_LOG_DEBUG, "%04x ",i); av_log(NULL, AV_LOG_DEBUG, "%02x ",buf[i]); if ((i&15)==15) av_log(NULL, AV_LOG_DEBUG, "\n"); } av_log(NULL, AV_LOG_ERROR, "\n"); } #endif static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, 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; } #ifdef CONFIG_ENCODERS 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, }; #endif //CONFIG_ENCODERS AVCodec adpcm_adx_decoder = { "adpcm_adx", CODEC_TYPE_AUDIO, CODEC_ID_ADPCM_ADX, sizeof(ADXContext), adx_decode_init, NULL, NULL, adx_decode_frame, };