Mercurial > libavcodec.hg
view ws-snd1.c @ 7724:ea9aa2aa4caa libavcodec
dca: Do float -> int16 interleaving in-place using s->dsp.float_to_int16_interleave()
author | andoma |
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date | Fri, 29 Aug 2008 10:09:51 +0000 |
parents | 85ab7655ad4d |
children | 2acf0ae7b041 |
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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 av_cold int ws_snd_decode_init(AVCodecContext * avctx) { // WSSNDContext *c = avctx->priv_data; avctx->sample_fmt = SAMPLE_FMT_S16; return 0; } static int ws_snd_decode_frame(AVCodecContext *avctx, void *data, int *data_size, const 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 (out_size > *data_size) { av_log(avctx, AV_LOG_ERROR, "Frame is too large to fit in buffer\n"); return -1; } if (in_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n"); return -1; } 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, .long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"), };