Mercurial > libavcodec.hg
view ws-snd1.c @ 11032:01bd040f8607 libavcodec
Unroll main loop so the edge==0 case is seperate.
This allows many things to be simplified away.
h264 decoder is overall 1% faster with a mbaff sample and
0.1% slower with the cathedral sample, probably because the slow loop
filter code must be loaded into the code cache for each first MB of each
row but isnt used for the following MBs.
author | michael |
---|---|
date | Thu, 28 Jan 2010 01:24:25 +0000 |
parents | 54bc8a2727b0 |
children | 8a4984c5cacc |
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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 "libavutil/intreadwrite.h" #include "avcodec.h" /** * @file libavcodec/ws-snd1.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, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->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)"), };