Mercurial > libavcodec.hg
view dtsdec.c @ 4551:c0bf618fbe7e libavcodec
Add support for MS-GSM codec
author | mbardiaux |
---|---|
date | Tue, 20 Feb 2007 11:09:47 +0000 |
parents | ee7422a921cb |
children | 8f47dc8782f9 |
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/* * dtsdec.c : free DTS Coherent Acoustics stream decoder. * Copyright (C) 2004 Benjamin Zores <ben@geexbox.org> * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 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 General Public License for more details. * * You should have received a copy of the GNU 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 <dts.h> #include <stdlib.h> #include <string.h> #define BUFFER_SIZE 18726 #define HEADER_SIZE 14 #define CONVERT_LEVEL 1 #define CONVERT_BIAS 0 typedef struct DTSContext { dts_state_t *state; uint8_t buf[BUFFER_SIZE]; uint8_t *bufptr; uint8_t *bufpos; } DTSContext; static inline int16_t convert(sample_t s) { return s * 0x7fff; } static void convert2s16_multi(sample_t *f, int16_t *s16, int flags) { int i; switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){ case DTS_MONO: for(i = 0; i < 256; i++){ s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0; s16[5*i+4] = convert(f[i]); } case DTS_CHANNEL: case DTS_STEREO: case DTS_DOLBY: for(i = 0; i < 256; i++){ s16[2*i] = convert(f[i]); s16[2*i+1] = convert(f[i+256]); } case DTS_3F: for(i = 0; i < 256; i++){ s16[5*i] = convert(f[i+256]); s16[5*i+1] = convert(f[i+512]); s16[5*i+2] = s16[5*i+3] = 0; s16[5*i+4] = convert(f[i]); } case DTS_2F2R: for(i = 0; i < 256; i++){ s16[4*i] = convert(f[i]); s16[4*i+1] = convert(f[i+256]); s16[4*i+2] = convert(f[i+512]); s16[4*i+3] = convert(f[i+768]); } case DTS_3F2R: for(i = 0; i < 256; i++){ s16[5*i] = convert(f[i+256]); s16[5*i+1] = convert(f[i+512]); s16[5*i+2] = convert(f[i+768]); s16[5*i+3] = convert(f[i+1024]); s16[5*i+4] = convert(f[i]); } case DTS_MONO | DTS_LFE: for(i = 0; i < 256; i++){ s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0; s16[6*i+4] = convert(f[i]); s16[6*i+5] = convert(f[i+256]); } case DTS_CHANNEL | DTS_LFE: case DTS_STEREO | DTS_LFE: case DTS_DOLBY | DTS_LFE: for(i = 0; i < 256; i++){ s16[6*i] = convert(f[i]); s16[6*i+1] = convert(f[i+256]); s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0; s16[6*i+5] = convert(f[i+512]); } case DTS_3F | DTS_LFE: for(i = 0; i < 256; i++){ s16[6*i] = convert(f[i+256]); s16[6*i+1] = convert(f[i+512]); s16[6*i+2] = s16[6*i+3] = 0; s16[6*i+4] = convert(f[i]); s16[6*i+5] = convert(f[i+768]); } case DTS_2F2R | DTS_LFE: for(i = 0; i < 256; i++){ s16[6*i] = convert(f[i]); s16[6*i+1] = convert(f[i+256]); s16[6*i+2] = convert(f[i+512]); s16[6*i+3] = convert(f[i+768]); s16[6*i+4] = 0; s16[6*i+5] = convert(f[i+1024]); } case DTS_3F2R | DTS_LFE: for(i = 0; i < 256; i++){ s16[6*i] = convert(f[i+256]); s16[6*i+1] = convert(f[i+512]); s16[6*i+2] = convert(f[i+768]); s16[6*i+3] = convert(f[i+1024]); s16[6*i+4] = convert(f[i]); s16[6*i+5] = convert(f[i+1280]); } } } static int channels_multi(int flags) { switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){ case DTS_CHANNEL: case DTS_STEREO: case DTS_DOLBY: return 2; case DTS_2F2R: return 4; case DTS_MONO: case DTS_3F: case DTS_3F2R: return 5; case DTS_MONO | DTS_LFE: case DTS_CHANNEL | DTS_LFE: case DTS_STEREO | DTS_LFE: case DTS_DOLBY | DTS_LFE: case DTS_3F | DTS_LFE: case DTS_2F2R | DTS_LFE: case DTS_3F2R | DTS_LFE: return 6; } return -1; } static int dts_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t * buff, int buff_size) { DTSContext *s = avctx->priv_data; uint8_t *start = buff; uint8_t *end = buff + buff_size; int16_t *out_samples = data; int sample_rate; int frame_length; int flags; int bit_rate; int len; level_t level; sample_t bias; int nblocks; int i; *data_size = 0; while(1) { int length; len = end - start; if(!len) break; if(len > s->bufpos - s->bufptr) len = s->bufpos - s->bufptr; memcpy(s->bufptr, start, len); s->bufptr += len; start += len; if(s->bufptr != s->bufpos) return start - buff; if(s->bufpos != s->buf + HEADER_SIZE) break; length = dts_syncinfo(s->state, s->buf, &flags, &sample_rate, &bit_rate, &frame_length); if(!length) { av_log(NULL, AV_LOG_INFO, "skip\n"); for(s->bufptr = s->buf; s->bufptr < s->buf + HEADER_SIZE - 1; s->bufptr++) s->bufptr[0] = s->bufptr[1]; continue; } s->bufpos = s->buf + length; } level = CONVERT_LEVEL; bias = CONVERT_BIAS; flags |= DTS_ADJUST_LEVEL; if(dts_frame(s->state, s->buf, &flags, &level, bias)) { av_log(avctx, AV_LOG_ERROR, "dts_frame() failed\n"); goto end; } avctx->sample_rate = sample_rate; avctx->channels = channels_multi(flags); avctx->bit_rate = bit_rate; nblocks = dts_blocks_num(s->state); for(i = 0; i < nblocks; i++) { if(dts_block(s->state)) { av_log(avctx, AV_LOG_ERROR, "dts_block() failed\n"); goto end; } convert2s16_multi(dts_samples(s->state), out_samples, flags); out_samples += 256 * avctx->channels; *data_size += 256 * sizeof(int16_t) * avctx->channels; } end: s->bufptr = s->buf; s->bufpos = s->buf + HEADER_SIZE; return start - buff; } static int dts_decode_init(AVCodecContext * avctx) { DTSContext *s = avctx->priv_data; s->bufptr = s->buf; s->bufpos = s->buf + HEADER_SIZE; s->state = dts_init(0); if(s->state == NULL) return -1; return 0; } static int dts_decode_end(AVCodecContext * avctx) { DTSContext *s = avctx->priv_data; dts_free(s->state); return 0; } AVCodec dts_decoder = { "dts", CODEC_TYPE_AUDIO, CODEC_ID_DTS, sizeof(DTSContext), dts_decode_init, NULL, dts_decode_end, dts_decode_frame, };