mirror of
https://github.com/xenia-project/FFmpeg.git
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2912e87a6c
Signed-off-by: Mans Rullgard <mans@mansr.com>
1774 lines
65 KiB
C
1774 lines
65 KiB
C
/*
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* ADPCM codecs
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* Copyright (c) 2001-2003 The ffmpeg Project
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "get_bits.h"
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#include "put_bits.h"
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#include "bytestream.h"
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/**
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* @file
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* ADPCM codecs.
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* First version by Francois Revol (revol@free.fr)
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* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
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* by Mike Melanson (melanson@pcisys.net)
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* CD-ROM XA ADPCM codec by BERO
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* EA ADPCM decoder by Robin Kay (komadori@myrealbox.com)
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* EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org)
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* EA IMA EACS decoder by Peter Ross (pross@xvid.org)
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* EA IMA SEAD decoder by Peter Ross (pross@xvid.org)
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* EA ADPCM XAS decoder by Peter Ross (pross@xvid.org)
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* MAXIS EA ADPCM decoder by Robert Marston (rmarston@gmail.com)
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* THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl)
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*
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* Features and limitations:
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*
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* Reference documents:
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* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
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* http://www.geocities.com/SiliconValley/8682/aud3.txt
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* http://openquicktime.sourceforge.net/plugins.htm
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* XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html
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* http://www.cs.ucla.edu/~leec/mediabench/applications.html
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* SoX source code http://home.sprynet.com/~cbagwell/sox.html
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*
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* CD-ROM XA:
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* http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html
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* vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html
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* readstr http://www.geocities.co.jp/Playtown/2004/
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*/
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#define BLKSIZE 1024
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/* step_table[] and index_table[] are from the ADPCM reference source */
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/* This is the index table: */
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static const int index_table[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8,
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};
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/**
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* This is the step table. Note that many programs use slight deviations from
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* this table, but such deviations are negligible:
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*/
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static const int step_table[89] = {
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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/* These are for MS-ADPCM */
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/* AdaptationTable[], AdaptCoeff1[], and AdaptCoeff2[] are from libsndfile */
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static const int AdaptationTable[] = {
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230, 230, 230, 230, 307, 409, 512, 614,
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768, 614, 512, 409, 307, 230, 230, 230
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};
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/** Divided by 4 to fit in 8-bit integers */
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static const uint8_t AdaptCoeff1[] = {
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64, 128, 0, 48, 60, 115, 98
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};
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/** Divided by 4 to fit in 8-bit integers */
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static const int8_t AdaptCoeff2[] = {
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0, -64, 0, 16, 0, -52, -58
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};
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/* These are for CD-ROM XA ADPCM */
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static const int xa_adpcm_table[5][2] = {
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{ 0, 0 },
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{ 60, 0 },
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{ 115, -52 },
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{ 98, -55 },
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{ 122, -60 }
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};
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static const int ea_adpcm_table[] = {
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0, 240, 460, 392, 0, 0, -208, -220, 0, 1,
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3, 4, 7, 8, 10, 11, 0, -1, -3, -4
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};
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// padded to zero where table size is less then 16
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static const int swf_index_tables[4][16] = {
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/*2*/ { -1, 2 },
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/*3*/ { -1, -1, 2, 4 },
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/*4*/ { -1, -1, -1, -1, 2, 4, 6, 8 },
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/*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 }
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};
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static const int yamaha_indexscale[] = {
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230, 230, 230, 230, 307, 409, 512, 614,
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230, 230, 230, 230, 307, 409, 512, 614
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};
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static const int yamaha_difflookup[] = {
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1, 3, 5, 7, 9, 11, 13, 15,
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-1, -3, -5, -7, -9, -11, -13, -15
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};
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/* end of tables */
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typedef struct ADPCMChannelStatus {
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int predictor;
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short int step_index;
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int step;
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/* for encoding */
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int prev_sample;
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/* MS version */
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short sample1;
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short sample2;
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int coeff1;
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int coeff2;
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int idelta;
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} ADPCMChannelStatus;
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typedef struct TrellisPath {
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int nibble;
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int prev;
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} TrellisPath;
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typedef struct TrellisNode {
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uint32_t ssd;
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int path;
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int sample1;
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int sample2;
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int step;
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} TrellisNode;
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typedef struct ADPCMContext {
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ADPCMChannelStatus status[6];
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TrellisPath *paths;
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TrellisNode *node_buf;
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TrellisNode **nodep_buf;
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uint8_t *trellis_hash;
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} ADPCMContext;
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#define FREEZE_INTERVAL 128
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/* XXX: implement encoding */
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#if CONFIG_ENCODERS
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static av_cold int adpcm_encode_init(AVCodecContext *avctx)
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{
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ADPCMContext *s = avctx->priv_data;
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uint8_t *extradata;
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int i;
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if (avctx->channels > 2)
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return -1; /* only stereo or mono =) */
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if(avctx->trellis && (unsigned)avctx->trellis > 16U){
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av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
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return -1;
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}
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if (avctx->trellis) {
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int frontier = 1 << avctx->trellis;
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int max_paths = frontier * FREEZE_INTERVAL;
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FF_ALLOC_OR_GOTO(avctx, s->paths, max_paths * sizeof(*s->paths), error);
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FF_ALLOC_OR_GOTO(avctx, s->node_buf, 2 * frontier * sizeof(*s->node_buf), error);
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FF_ALLOC_OR_GOTO(avctx, s->nodep_buf, 2 * frontier * sizeof(*s->nodep_buf), error);
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FF_ALLOC_OR_GOTO(avctx, s->trellis_hash, 65536 * sizeof(*s->trellis_hash), error);
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}
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switch(avctx->codec->id) {
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case CODEC_ID_ADPCM_IMA_WAV:
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avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 / (4 * avctx->channels) + 1; /* each 16 bits sample gives one nibble */
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/* and we have 4 bytes per channel overhead */
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avctx->block_align = BLKSIZE;
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/* seems frame_size isn't taken into account... have to buffer the samples :-( */
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break;
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case CODEC_ID_ADPCM_IMA_QT:
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avctx->frame_size = 64;
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avctx->block_align = 34 * avctx->channels;
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break;
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case CODEC_ID_ADPCM_MS:
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avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */
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/* and we have 7 bytes per channel overhead */
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avctx->block_align = BLKSIZE;
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avctx->extradata_size = 32;
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extradata = avctx->extradata = av_malloc(avctx->extradata_size);
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if (!extradata)
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return AVERROR(ENOMEM);
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bytestream_put_le16(&extradata, avctx->frame_size);
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bytestream_put_le16(&extradata, 7); /* wNumCoef */
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for (i = 0; i < 7; i++) {
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bytestream_put_le16(&extradata, AdaptCoeff1[i] * 4);
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bytestream_put_le16(&extradata, AdaptCoeff2[i] * 4);
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}
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break;
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case CODEC_ID_ADPCM_YAMAHA:
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avctx->frame_size = BLKSIZE * avctx->channels;
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avctx->block_align = BLKSIZE;
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break;
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case CODEC_ID_ADPCM_SWF:
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if (avctx->sample_rate != 11025 &&
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avctx->sample_rate != 22050 &&
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avctx->sample_rate != 44100) {
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av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, 22050 or 44100\n");
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goto error;
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}
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avctx->frame_size = 512 * (avctx->sample_rate / 11025);
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break;
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default:
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goto error;
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}
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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return 0;
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error:
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av_freep(&s->paths);
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av_freep(&s->node_buf);
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av_freep(&s->nodep_buf);
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av_freep(&s->trellis_hash);
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return -1;
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}
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static av_cold int adpcm_encode_close(AVCodecContext *avctx)
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{
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ADPCMContext *s = avctx->priv_data;
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av_freep(&avctx->coded_frame);
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av_freep(&s->paths);
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av_freep(&s->node_buf);
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av_freep(&s->nodep_buf);
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av_freep(&s->trellis_hash);
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return 0;
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}
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static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, short sample)
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{
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int delta = sample - c->prev_sample;
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int nibble = FFMIN(7, abs(delta)*4/step_table[c->step_index]) + (delta<0)*8;
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c->prev_sample += ((step_table[c->step_index] * yamaha_difflookup[nibble]) / 8);
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c->prev_sample = av_clip_int16(c->prev_sample);
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c->step_index = av_clip(c->step_index + index_table[nibble], 0, 88);
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return nibble;
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}
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static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, short sample)
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{
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int predictor, nibble, bias;
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predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
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nibble= sample - predictor;
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if(nibble>=0) bias= c->idelta/2;
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else bias=-c->idelta/2;
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nibble= (nibble + bias) / c->idelta;
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nibble= av_clip(nibble, -8, 7)&0x0F;
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predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
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c->sample2 = c->sample1;
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c->sample1 = av_clip_int16(predictor);
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c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8;
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if (c->idelta < 16) c->idelta = 16;
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return nibble;
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}
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static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, short sample)
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{
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int nibble, delta;
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if(!c->step) {
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c->predictor = 0;
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c->step = 127;
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}
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delta = sample - c->predictor;
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nibble = FFMIN(7, abs(delta)*4/c->step) + (delta<0)*8;
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c->predictor += ((c->step * yamaha_difflookup[nibble]) / 8);
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c->predictor = av_clip_int16(c->predictor);
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c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
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c->step = av_clip(c->step, 127, 24567);
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return nibble;
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}
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static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
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uint8_t *dst, ADPCMChannelStatus *c, int n)
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{
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//FIXME 6% faster if frontier is a compile-time constant
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ADPCMContext *s = avctx->priv_data;
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const int frontier = 1 << avctx->trellis;
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const int stride = avctx->channels;
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const int version = avctx->codec->id;
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TrellisPath *paths = s->paths, *p;
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TrellisNode *node_buf = s->node_buf;
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TrellisNode **nodep_buf = s->nodep_buf;
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TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
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TrellisNode **nodes_next = nodep_buf + frontier;
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int pathn = 0, froze = -1, i, j, k, generation = 0;
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uint8_t *hash = s->trellis_hash;
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memset(hash, 0xff, 65536 * sizeof(*hash));
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memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
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nodes[0] = node_buf + frontier;
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nodes[0]->ssd = 0;
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nodes[0]->path = 0;
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nodes[0]->step = c->step_index;
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nodes[0]->sample1 = c->sample1;
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nodes[0]->sample2 = c->sample2;
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if((version == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_IMA_QT) || (version == CODEC_ID_ADPCM_SWF))
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nodes[0]->sample1 = c->prev_sample;
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if(version == CODEC_ID_ADPCM_MS)
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nodes[0]->step = c->idelta;
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if(version == CODEC_ID_ADPCM_YAMAHA) {
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if(c->step == 0) {
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nodes[0]->step = 127;
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nodes[0]->sample1 = 0;
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} else {
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nodes[0]->step = c->step;
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nodes[0]->sample1 = c->predictor;
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}
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}
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for(i=0; i<n; i++) {
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TrellisNode *t = node_buf + frontier*(i&1);
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TrellisNode **u;
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int sample = samples[i*stride];
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int heap_pos = 0;
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memset(nodes_next, 0, frontier*sizeof(TrellisNode*));
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for(j=0; j<frontier && nodes[j]; j++) {
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// higher j have higher ssd already, so they're likely to yield a suboptimal next sample too
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const int range = (j < frontier/2) ? 1 : 0;
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const int step = nodes[j]->step;
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int nidx;
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if(version == CODEC_ID_ADPCM_MS) {
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const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 64;
|
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const int div = (sample - predictor) / step;
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const int nmin = av_clip(div-range, -8, 6);
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const int nmax = av_clip(div+range, -7, 7);
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for(nidx=nmin; nidx<=nmax; nidx++) {
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const int nibble = nidx & 0xf;
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int dec_sample = predictor + nidx * step;
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#define STORE_NODE(NAME, STEP_INDEX)\
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int d;\
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uint32_t ssd;\
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int pos;\
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TrellisNode *u;\
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uint8_t *h;\
|
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dec_sample = av_clip_int16(dec_sample);\
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d = sample - dec_sample;\
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ssd = nodes[j]->ssd + d*d;\
|
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/* Check for wraparound, skip such samples completely. \
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* Note, changing ssd to a 64 bit variable would be \
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* simpler, avoiding this check, but it's slower on \
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* x86 32 bit at the moment. */\
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if (ssd < nodes[j]->ssd)\
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goto next_##NAME;\
|
|
/* Collapse any two states with the same previous sample value. \
|
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* One could also distinguish states by step and by 2nd to last
|
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* sample, but the effects of that are negligible.
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* Since nodes in the previous generation are iterated
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* through a heap, they're roughly ordered from better to
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* worse, but not strictly ordered. Therefore, an earlier
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* node with the same sample value is better in most cases
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* (and thus the current is skipped), but not strictly
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* in all cases. Only skipping samples where ssd >=
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* ssd of the earlier node with the same sample gives
|
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* slightly worse quality, though, for some reason. */ \
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h = &hash[(uint16_t) dec_sample];\
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if (*h == generation)\
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goto next_##NAME;\
|
|
if (heap_pos < frontier) {\
|
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pos = heap_pos++;\
|
|
} else {\
|
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/* Try to replace one of the leaf nodes with the new \
|
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* one, but try a different slot each time. */\
|
|
pos = (frontier >> 1) + (heap_pos & ((frontier >> 1) - 1));\
|
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if (ssd > nodes_next[pos]->ssd)\
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goto next_##NAME;\
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heap_pos++;\
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}\
|
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*h = generation;\
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u = nodes_next[pos];\
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if(!u) {\
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assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\
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u = t++;\
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nodes_next[pos] = u;\
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u->path = pathn++;\
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}\
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u->ssd = ssd;\
|
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u->step = STEP_INDEX;\
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u->sample2 = nodes[j]->sample1;\
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u->sample1 = dec_sample;\
|
|
paths[u->path].nibble = nibble;\
|
|
paths[u->path].prev = nodes[j]->path;\
|
|
/* Sift the newly inserted node up in the heap to \
|
|
* restore the heap property. */\
|
|
while (pos > 0) {\
|
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int parent = (pos - 1) >> 1;\
|
|
if (nodes_next[parent]->ssd <= ssd)\
|
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break;\
|
|
FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
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pos = parent;\
|
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}\
|
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next_##NAME:;
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|
STORE_NODE(ms, FFMAX(16, (AdaptationTable[nibble] * step) >> 8));
|
|
}
|
|
} else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (version == CODEC_ID_ADPCM_IMA_QT)|| (version == CODEC_ID_ADPCM_SWF)) {
|
|
#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
|
|
const int predictor = nodes[j]->sample1;\
|
|
const int div = (sample - predictor) * 4 / STEP_TABLE;\
|
|
int nmin = av_clip(div-range, -7, 6);\
|
|
int nmax = av_clip(div+range, -6, 7);\
|
|
if(nmin<=0) nmin--; /* distinguish -0 from +0 */\
|
|
if(nmax<0) nmax--;\
|
|
for(nidx=nmin; nidx<=nmax; nidx++) {\
|
|
const int nibble = nidx<0 ? 7-nidx : nidx;\
|
|
int dec_sample = predictor + (STEP_TABLE * yamaha_difflookup[nibble]) / 8;\
|
|
STORE_NODE(NAME, STEP_INDEX);\
|
|
}
|
|
LOOP_NODES(ima, step_table[step], av_clip(step + index_table[nibble], 0, 88));
|
|
} else { //CODEC_ID_ADPCM_YAMAHA
|
|
LOOP_NODES(yamaha, step, av_clip((step * yamaha_indexscale[nibble]) >> 8, 127, 24567));
|
|
#undef LOOP_NODES
|
|
#undef STORE_NODE
|
|
}
|
|
}
|
|
|
|
u = nodes;
|
|
nodes = nodes_next;
|
|
nodes_next = u;
|
|
|
|
generation++;
|
|
if (generation == 255) {
|
|
memset(hash, 0xff, 65536 * sizeof(*hash));
|
|
generation = 0;
|
|
}
|
|
|
|
// prevent overflow
|
|
if(nodes[0]->ssd > (1<<28)) {
|
|
for(j=1; j<frontier && nodes[j]; j++)
|
|
nodes[j]->ssd -= nodes[0]->ssd;
|
|
nodes[0]->ssd = 0;
|
|
}
|
|
|
|
// merge old paths to save memory
|
|
if(i == froze + FREEZE_INTERVAL) {
|
|
p = &paths[nodes[0]->path];
|
|
for(k=i; k>froze; k--) {
|
|
dst[k] = p->nibble;
|
|
p = &paths[p->prev];
|
|
}
|
|
froze = i;
|
|
pathn = 0;
|
|
// other nodes might use paths that don't coincide with the frozen one.
|
|
// checking which nodes do so is too slow, so just kill them all.
|
|
// this also slightly improves quality, but I don't know why.
|
|
memset(nodes+1, 0, (frontier-1)*sizeof(TrellisNode*));
|
|
}
|
|
}
|
|
|
|
p = &paths[nodes[0]->path];
|
|
for(i=n-1; i>froze; i--) {
|
|
dst[i] = p->nibble;
|
|
p = &paths[p->prev];
|
|
}
|
|
|
|
c->predictor = nodes[0]->sample1;
|
|
c->sample1 = nodes[0]->sample1;
|
|
c->sample2 = nodes[0]->sample2;
|
|
c->step_index = nodes[0]->step;
|
|
c->step = nodes[0]->step;
|
|
c->idelta = nodes[0]->step;
|
|
}
|
|
|
|
static int adpcm_encode_frame(AVCodecContext *avctx,
|
|
unsigned char *frame, int buf_size, void *data)
|
|
{
|
|
int n, i, st;
|
|
short *samples;
|
|
unsigned char *dst;
|
|
ADPCMContext *c = avctx->priv_data;
|
|
uint8_t *buf;
|
|
|
|
dst = frame;
|
|
samples = (short *)data;
|
|
st= avctx->channels == 2;
|
|
/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
|
|
|
|
switch(avctx->codec->id) {
|
|
case CODEC_ID_ADPCM_IMA_WAV:
|
|
n = avctx->frame_size / 8;
|
|
c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
|
|
/* c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */
|
|
bytestream_put_le16(&dst, c->status[0].prev_sample);
|
|
*dst++ = (unsigned char)c->status[0].step_index;
|
|
*dst++ = 0; /* unknown */
|
|
samples++;
|
|
if (avctx->channels == 2) {
|
|
c->status[1].prev_sample = (signed short)samples[0];
|
|
/* c->status[1].step_index = 0; */
|
|
bytestream_put_le16(&dst, c->status[1].prev_sample);
|
|
*dst++ = (unsigned char)c->status[1].step_index;
|
|
*dst++ = 0;
|
|
samples++;
|
|
}
|
|
|
|
/* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */
|
|
if(avctx->trellis > 0) {
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error);
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8);
|
|
if(avctx->channels == 2)
|
|
adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8);
|
|
for(i=0; i<n; i++) {
|
|
*dst++ = buf[8*i+0] | (buf[8*i+1] << 4);
|
|
*dst++ = buf[8*i+2] | (buf[8*i+3] << 4);
|
|
*dst++ = buf[8*i+4] | (buf[8*i+5] << 4);
|
|
*dst++ = buf[8*i+6] | (buf[8*i+7] << 4);
|
|
if (avctx->channels == 2) {
|
|
uint8_t *buf1 = buf + n*8;
|
|
*dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4);
|
|
*dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4);
|
|
*dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4);
|
|
*dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4);
|
|
}
|
|
}
|
|
av_free(buf);
|
|
} else
|
|
for (; n>0; n--) {
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
|
|
dst++;
|
|
/* right channel */
|
|
if (avctx->channels == 2) {
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[1]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[5]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[9]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
|
|
dst++;
|
|
}
|
|
samples += 8 * avctx->channels;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_QT:
|
|
{
|
|
int ch, i;
|
|
PutBitContext pb;
|
|
init_put_bits(&pb, dst, buf_size*8);
|
|
|
|
for(ch=0; ch<avctx->channels; ch++){
|
|
put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
|
|
put_bits(&pb, 7, c->status[ch].step_index);
|
|
if(avctx->trellis > 0) {
|
|
uint8_t buf[64];
|
|
adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
|
|
for(i=0; i<64; i++)
|
|
put_bits(&pb, 4, buf[i^1]);
|
|
c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F;
|
|
} else {
|
|
for (i=0; i<64; i+=2){
|
|
int t1, t2;
|
|
t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
|
|
t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
|
|
put_bits(&pb, 4, t2);
|
|
put_bits(&pb, 4, t1);
|
|
}
|
|
c->status[ch].prev_sample &= ~0x7F;
|
|
}
|
|
}
|
|
|
|
flush_put_bits(&pb);
|
|
dst += put_bits_count(&pb)>>3;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_SWF:
|
|
{
|
|
int i;
|
|
PutBitContext pb;
|
|
init_put_bits(&pb, dst, buf_size*8);
|
|
|
|
n = avctx->frame_size-1;
|
|
|
|
//Store AdpcmCodeSize
|
|
put_bits(&pb, 2, 2); //Set 4bits flash adpcm format
|
|
|
|
//Init the encoder state
|
|
for(i=0; i<avctx->channels; i++){
|
|
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits
|
|
put_sbits(&pb, 16, samples[i]);
|
|
put_bits(&pb, 6, c->status[i].step_index);
|
|
c->status[i].prev_sample = (signed short)samples[i];
|
|
}
|
|
|
|
if(avctx->trellis > 0) {
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
|
|
adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n);
|
|
if (avctx->channels == 2)
|
|
adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n);
|
|
for(i=0; i<n; i++) {
|
|
put_bits(&pb, 4, buf[i]);
|
|
if (avctx->channels == 2)
|
|
put_bits(&pb, 4, buf[n+i]);
|
|
}
|
|
av_free(buf);
|
|
} else {
|
|
for (i=1; i<avctx->frame_size; i++) {
|
|
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i]));
|
|
if (avctx->channels == 2)
|
|
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1]));
|
|
}
|
|
}
|
|
flush_put_bits(&pb);
|
|
dst += put_bits_count(&pb)>>3;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_MS:
|
|
for(i=0; i<avctx->channels; i++){
|
|
int predictor=0;
|
|
|
|
*dst++ = predictor;
|
|
c->status[i].coeff1 = AdaptCoeff1[predictor];
|
|
c->status[i].coeff2 = AdaptCoeff2[predictor];
|
|
}
|
|
for(i=0; i<avctx->channels; i++){
|
|
if (c->status[i].idelta < 16)
|
|
c->status[i].idelta = 16;
|
|
|
|
bytestream_put_le16(&dst, c->status[i].idelta);
|
|
}
|
|
for(i=0; i<avctx->channels; i++){
|
|
c->status[i].sample2= *samples++;
|
|
}
|
|
for(i=0; i<avctx->channels; i++){
|
|
c->status[i].sample1= *samples++;
|
|
|
|
bytestream_put_le16(&dst, c->status[i].sample1);
|
|
}
|
|
for(i=0; i<avctx->channels; i++)
|
|
bytestream_put_le16(&dst, c->status[i].sample2);
|
|
|
|
if(avctx->trellis > 0) {
|
|
int n = avctx->block_align - 7*avctx->channels;
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
|
|
if(avctx->channels == 1) {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
for(i=0; i<n; i+=2)
|
|
*dst++ = (buf[i] << 4) | buf[i+1];
|
|
} else {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
|
|
for(i=0; i<n; i++)
|
|
*dst++ = (buf[i] << 4) | buf[n+i];
|
|
}
|
|
av_free(buf);
|
|
} else
|
|
for(i=7*avctx->channels; i<avctx->block_align; i++) {
|
|
int nibble;
|
|
nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++)<<4;
|
|
nibble|= adpcm_ms_compress_sample(&c->status[st], *samples++);
|
|
*dst++ = nibble;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_YAMAHA:
|
|
n = avctx->frame_size / 2;
|
|
if(avctx->trellis > 0) {
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error);
|
|
n *= 2;
|
|
if(avctx->channels == 1) {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
for(i=0; i<n; i+=2)
|
|
*dst++ = buf[i] | (buf[i+1] << 4);
|
|
} else {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
|
|
for(i=0; i<n; i++)
|
|
*dst++ = buf[i] | (buf[n+i] << 4);
|
|
}
|
|
av_free(buf);
|
|
} else
|
|
for (n *= avctx->channels; n>0; n--) {
|
|
int nibble;
|
|
nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
|
|
nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
|
|
*dst++ = nibble;
|
|
}
|
|
break;
|
|
default:
|
|
error:
|
|
return -1;
|
|
}
|
|
return dst - frame;
|
|
}
|
|
#endif //CONFIG_ENCODERS
|
|
|
|
static av_cold int adpcm_decode_init(AVCodecContext * avctx)
|
|
{
|
|
ADPCMContext *c = avctx->priv_data;
|
|
unsigned int max_channels = 2;
|
|
|
|
switch(avctx->codec->id) {
|
|
case CODEC_ID_ADPCM_EA_R1:
|
|
case CODEC_ID_ADPCM_EA_R2:
|
|
case CODEC_ID_ADPCM_EA_R3:
|
|
max_channels = 6;
|
|
break;
|
|
}
|
|
if(avctx->channels > max_channels){
|
|
return -1;
|
|
}
|
|
|
|
switch(avctx->codec->id) {
|
|
case CODEC_ID_ADPCM_CT:
|
|
c->status[0].step = c->status[1].step = 511;
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WAV:
|
|
if (avctx->bits_per_coded_sample != 4) {
|
|
av_log(avctx, AV_LOG_ERROR, "Only 4-bit ADPCM IMA WAV files are supported\n");
|
|
return -1;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WS:
|
|
if (avctx->extradata && avctx->extradata_size == 2 * 4) {
|
|
c->status[0].predictor = AV_RL32(avctx->extradata);
|
|
c->status[1].predictor = AV_RL32(avctx->extradata + 4);
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
|
|
return 0;
|
|
}
|
|
|
|
static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble, int shift)
|
|
{
|
|
int step_index;
|
|
int predictor;
|
|
int sign, delta, diff, step;
|
|
|
|
step = step_table[c->step_index];
|
|
step_index = c->step_index + index_table[(unsigned)nibble];
|
|
if (step_index < 0) step_index = 0;
|
|
else if (step_index > 88) step_index = 88;
|
|
|
|
sign = nibble & 8;
|
|
delta = nibble & 7;
|
|
/* perform direct multiplication instead of series of jumps proposed by
|
|
* the reference ADPCM implementation since modern CPUs can do the mults
|
|
* quickly enough */
|
|
diff = ((2 * delta + 1) * step) >> shift;
|
|
predictor = c->predictor;
|
|
if (sign) predictor -= diff;
|
|
else predictor += diff;
|
|
|
|
c->predictor = av_clip_int16(predictor);
|
|
c->step_index = step_index;
|
|
|
|
return (short)c->predictor;
|
|
}
|
|
|
|
static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
|
|
{
|
|
int predictor;
|
|
|
|
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
|
|
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
|
|
|
|
c->sample2 = c->sample1;
|
|
c->sample1 = av_clip_int16(predictor);
|
|
c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8;
|
|
if (c->idelta < 16) c->idelta = 16;
|
|
|
|
return c->sample1;
|
|
}
|
|
|
|
static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
|
|
{
|
|
int sign, delta, diff;
|
|
int new_step;
|
|
|
|
sign = nibble & 8;
|
|
delta = nibble & 7;
|
|
/* perform direct multiplication instead of series of jumps proposed by
|
|
* the reference ADPCM implementation since modern CPUs can do the mults
|
|
* quickly enough */
|
|
diff = ((2 * delta + 1) * c->step) >> 3;
|
|
/* predictor update is not so trivial: predictor is multiplied on 254/256 before updating */
|
|
c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff);
|
|
c->predictor = av_clip_int16(c->predictor);
|
|
/* calculate new step and clamp it to range 511..32767 */
|
|
new_step = (AdaptationTable[nibble & 7] * c->step) >> 8;
|
|
c->step = av_clip(new_step, 511, 32767);
|
|
|
|
return (short)c->predictor;
|
|
}
|
|
|
|
static inline short adpcm_sbpro_expand_nibble(ADPCMChannelStatus *c, char nibble, int size, int shift)
|
|
{
|
|
int sign, delta, diff;
|
|
|
|
sign = nibble & (1<<(size-1));
|
|
delta = nibble & ((1<<(size-1))-1);
|
|
diff = delta << (7 + c->step + shift);
|
|
|
|
/* clamp result */
|
|
c->predictor = av_clip(c->predictor + (sign ? -diff : diff), -16384,16256);
|
|
|
|
/* calculate new step */
|
|
if (delta >= (2*size - 3) && c->step < 3)
|
|
c->step++;
|
|
else if (delta == 0 && c->step > 0)
|
|
c->step--;
|
|
|
|
return (short) c->predictor;
|
|
}
|
|
|
|
static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned char nibble)
|
|
{
|
|
if(!c->step) {
|
|
c->predictor = 0;
|
|
c->step = 127;
|
|
}
|
|
|
|
c->predictor += (c->step * yamaha_difflookup[nibble]) / 8;
|
|
c->predictor = av_clip_int16(c->predictor);
|
|
c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
|
|
c->step = av_clip(c->step, 127, 24567);
|
|
return c->predictor;
|
|
}
|
|
|
|
static void xa_decode(short *out, const unsigned char *in,
|
|
ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
|
|
{
|
|
int i, j;
|
|
int shift,filter,f0,f1;
|
|
int s_1,s_2;
|
|
int d,s,t;
|
|
|
|
for(i=0;i<4;i++) {
|
|
|
|
shift = 12 - (in[4+i*2] & 15);
|
|
filter = in[4+i*2] >> 4;
|
|
f0 = xa_adpcm_table[filter][0];
|
|
f1 = xa_adpcm_table[filter][1];
|
|
|
|
s_1 = left->sample1;
|
|
s_2 = left->sample2;
|
|
|
|
for(j=0;j<28;j++) {
|
|
d = in[16+i+j*4];
|
|
|
|
t = (signed char)(d<<4)>>4;
|
|
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
|
|
s_2 = s_1;
|
|
s_1 = av_clip_int16(s);
|
|
*out = s_1;
|
|
out += inc;
|
|
}
|
|
|
|
if (inc==2) { /* stereo */
|
|
left->sample1 = s_1;
|
|
left->sample2 = s_2;
|
|
s_1 = right->sample1;
|
|
s_2 = right->sample2;
|
|
out = out + 1 - 28*2;
|
|
}
|
|
|
|
shift = 12 - (in[5+i*2] & 15);
|
|
filter = in[5+i*2] >> 4;
|
|
|
|
f0 = xa_adpcm_table[filter][0];
|
|
f1 = xa_adpcm_table[filter][1];
|
|
|
|
for(j=0;j<28;j++) {
|
|
d = in[16+i+j*4];
|
|
|
|
t = (signed char)d >> 4;
|
|
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
|
|
s_2 = s_1;
|
|
s_1 = av_clip_int16(s);
|
|
*out = s_1;
|
|
out += inc;
|
|
}
|
|
|
|
if (inc==2) { /* stereo */
|
|
right->sample1 = s_1;
|
|
right->sample2 = s_2;
|
|
out -= 1;
|
|
} else {
|
|
left->sample1 = s_1;
|
|
left->sample2 = s_2;
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/* DK3 ADPCM support macro */
|
|
#define DK3_GET_NEXT_NIBBLE() \
|
|
if (decode_top_nibble_next) \
|
|
{ \
|
|
nibble = last_byte >> 4; \
|
|
decode_top_nibble_next = 0; \
|
|
} \
|
|
else \
|
|
{ \
|
|
last_byte = *src++; \
|
|
if (src >= buf + buf_size) break; \
|
|
nibble = last_byte & 0x0F; \
|
|
decode_top_nibble_next = 1; \
|
|
}
|
|
|
|
static int adpcm_decode_frame(AVCodecContext *avctx,
|
|
void *data, int *data_size,
|
|
AVPacket *avpkt)
|
|
{
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
ADPCMContext *c = avctx->priv_data;
|
|
ADPCMChannelStatus *cs;
|
|
int n, m, channel, i;
|
|
int block_predictor[2];
|
|
short *samples;
|
|
short *samples_end;
|
|
const uint8_t *src;
|
|
int st; /* stereo */
|
|
|
|
/* DK3 ADPCM accounting variables */
|
|
unsigned char last_byte = 0;
|
|
unsigned char nibble;
|
|
int decode_top_nibble_next = 0;
|
|
int diff_channel;
|
|
|
|
/* EA ADPCM state variables */
|
|
uint32_t samples_in_chunk;
|
|
int32_t previous_left_sample, previous_right_sample;
|
|
int32_t current_left_sample, current_right_sample;
|
|
int32_t next_left_sample, next_right_sample;
|
|
int32_t coeff1l, coeff2l, coeff1r, coeff2r;
|
|
uint8_t shift_left, shift_right;
|
|
int count1, count2;
|
|
int coeff[2][2], shift[2];//used in EA MAXIS ADPCM
|
|
|
|
if (!buf_size)
|
|
return 0;
|
|
|
|
//should protect all 4bit ADPCM variants
|
|
//8 is needed for CODEC_ID_ADPCM_IMA_WAV with 2 channels
|
|
//
|
|
if(*data_size/4 < buf_size + 8)
|
|
return -1;
|
|
|
|
samples = data;
|
|
samples_end= samples + *data_size/2;
|
|
*data_size= 0;
|
|
src = buf;
|
|
|
|
st = avctx->channels == 2 ? 1 : 0;
|
|
|
|
switch(avctx->codec->id) {
|
|
case CODEC_ID_ADPCM_IMA_QT:
|
|
n = buf_size - 2*avctx->channels;
|
|
for (channel = 0; channel < avctx->channels; channel++) {
|
|
cs = &(c->status[channel]);
|
|
/* (pppppp) (piiiiiii) */
|
|
|
|
/* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
|
|
cs->predictor = (*src++) << 8;
|
|
cs->predictor |= (*src & 0x80);
|
|
cs->predictor &= 0xFF80;
|
|
|
|
/* sign extension */
|
|
if(cs->predictor & 0x8000)
|
|
cs->predictor -= 0x10000;
|
|
|
|
cs->predictor = av_clip_int16(cs->predictor);
|
|
|
|
cs->step_index = (*src++) & 0x7F;
|
|
|
|
if (cs->step_index > 88){
|
|
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
|
|
cs->step_index = 88;
|
|
}
|
|
|
|
cs->step = step_table[cs->step_index];
|
|
|
|
samples = (short*)data + channel;
|
|
|
|
for(m=32; n>0 && m>0; n--, m--) { /* in QuickTime, IMA is encoded by chuncks of 34 bytes (=64 samples) */
|
|
*samples = adpcm_ima_expand_nibble(cs, src[0] & 0x0F, 3);
|
|
samples += avctx->channels;
|
|
*samples = adpcm_ima_expand_nibble(cs, src[0] >> 4 , 3);
|
|
samples += avctx->channels;
|
|
src ++;
|
|
}
|
|
}
|
|
if (st)
|
|
samples--;
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WAV:
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
// samples_per_block= (block_align-4*chanels)*8 / (bits_per_sample * chanels) + 1;
|
|
|
|
for(i=0; i<avctx->channels; i++){
|
|
cs = &(c->status[i]);
|
|
cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src);
|
|
|
|
cs->step_index = *src++;
|
|
if (cs->step_index > 88){
|
|
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
|
|
cs->step_index = 88;
|
|
}
|
|
if (*src++) av_log(avctx, AV_LOG_ERROR, "unused byte should be null but is %d!!\n", src[-1]); /* unused */
|
|
}
|
|
|
|
while(src < buf + buf_size){
|
|
for(m=0; m<4; m++){
|
|
for(i=0; i<=st; i++)
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] & 0x0F, 3);
|
|
for(i=0; i<=st; i++)
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[i], src[4*i] >> 4 , 3);
|
|
src++;
|
|
}
|
|
src += 4*st;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_4XM:
|
|
cs = &(c->status[0]);
|
|
c->status[0].predictor= (int16_t)bytestream_get_le16(&src);
|
|
if(st){
|
|
c->status[1].predictor= (int16_t)bytestream_get_le16(&src);
|
|
}
|
|
c->status[0].step_index= (int16_t)bytestream_get_le16(&src);
|
|
if(st){
|
|
c->status[1].step_index= (int16_t)bytestream_get_le16(&src);
|
|
}
|
|
if (cs->step_index < 0) cs->step_index = 0;
|
|
if (cs->step_index > 88) cs->step_index = 88;
|
|
|
|
m= (buf_size - (src - buf))>>st;
|
|
for(i=0; i<m; i++) {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] & 0x0F, 4);
|
|
if (st)
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] & 0x0F, 4);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[i] >> 4, 4);
|
|
if (st)
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[1], src[i+m] >> 4, 4);
|
|
}
|
|
|
|
src += m<<st;
|
|
|
|
break;
|
|
case CODEC_ID_ADPCM_MS:
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
n = buf_size - 7 * avctx->channels;
|
|
if (n < 0)
|
|
return -1;
|
|
block_predictor[0] = av_clip(*src++, 0, 6);
|
|
block_predictor[1] = 0;
|
|
if (st)
|
|
block_predictor[1] = av_clip(*src++, 0, 6);
|
|
c->status[0].idelta = (int16_t)bytestream_get_le16(&src);
|
|
if (st){
|
|
c->status[1].idelta = (int16_t)bytestream_get_le16(&src);
|
|
}
|
|
c->status[0].coeff1 = AdaptCoeff1[block_predictor[0]];
|
|
c->status[0].coeff2 = AdaptCoeff2[block_predictor[0]];
|
|
c->status[1].coeff1 = AdaptCoeff1[block_predictor[1]];
|
|
c->status[1].coeff2 = AdaptCoeff2[block_predictor[1]];
|
|
|
|
c->status[0].sample1 = bytestream_get_le16(&src);
|
|
if (st) c->status[1].sample1 = bytestream_get_le16(&src);
|
|
c->status[0].sample2 = bytestream_get_le16(&src);
|
|
if (st) c->status[1].sample2 = bytestream_get_le16(&src);
|
|
|
|
*samples++ = c->status[0].sample2;
|
|
if (st) *samples++ = c->status[1].sample2;
|
|
*samples++ = c->status[0].sample1;
|
|
if (st) *samples++ = c->status[1].sample1;
|
|
for(;n>0;n--) {
|
|
*samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
|
|
*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
|
|
src ++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_DK4:
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
c->status[0].predictor = (int16_t)bytestream_get_le16(&src);
|
|
c->status[0].step_index = *src++;
|
|
src++;
|
|
*samples++ = c->status[0].predictor;
|
|
if (st) {
|
|
c->status[1].predictor = (int16_t)bytestream_get_le16(&src);
|
|
c->status[1].step_index = *src++;
|
|
src++;
|
|
*samples++ = c->status[1].predictor;
|
|
}
|
|
while (src < buf + buf_size) {
|
|
|
|
/* take care of the top nibble (always left or mono channel) */
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] >> 4, 3);
|
|
|
|
/* take care of the bottom nibble, which is right sample for
|
|
* stereo, or another mono sample */
|
|
if (st)
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
|
|
src[0] & 0x0F, 3);
|
|
else
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] & 0x0F, 3);
|
|
|
|
src++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_DK3:
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
if(buf_size + 16 > (samples_end - samples)*3/8)
|
|
return -1;
|
|
|
|
c->status[0].predictor = (int16_t)AV_RL16(src + 10);
|
|
c->status[1].predictor = (int16_t)AV_RL16(src + 12);
|
|
c->status[0].step_index = src[14];
|
|
c->status[1].step_index = src[15];
|
|
/* sign extend the predictors */
|
|
src += 16;
|
|
diff_channel = c->status[1].predictor;
|
|
|
|
/* the DK3_GET_NEXT_NIBBLE macro issues the break statement when
|
|
* the buffer is consumed */
|
|
while (1) {
|
|
|
|
/* for this algorithm, c->status[0] is the sum channel and
|
|
* c->status[1] is the diff channel */
|
|
|
|
/* process the first predictor of the sum channel */
|
|
DK3_GET_NEXT_NIBBLE();
|
|
adpcm_ima_expand_nibble(&c->status[0], nibble, 3);
|
|
|
|
/* process the diff channel predictor */
|
|
DK3_GET_NEXT_NIBBLE();
|
|
adpcm_ima_expand_nibble(&c->status[1], nibble, 3);
|
|
|
|
/* process the first pair of stereo PCM samples */
|
|
diff_channel = (diff_channel + c->status[1].predictor) / 2;
|
|
*samples++ = c->status[0].predictor + c->status[1].predictor;
|
|
*samples++ = c->status[0].predictor - c->status[1].predictor;
|
|
|
|
/* process the second predictor of the sum channel */
|
|
DK3_GET_NEXT_NIBBLE();
|
|
adpcm_ima_expand_nibble(&c->status[0], nibble, 3);
|
|
|
|
/* process the second pair of stereo PCM samples */
|
|
diff_channel = (diff_channel + c->status[1].predictor) / 2;
|
|
*samples++ = c->status[0].predictor + c->status[1].predictor;
|
|
*samples++ = c->status[0].predictor - c->status[1].predictor;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_ISS:
|
|
c->status[0].predictor = (int16_t)AV_RL16(src + 0);
|
|
c->status[0].step_index = src[2];
|
|
src += 4;
|
|
if(st) {
|
|
c->status[1].predictor = (int16_t)AV_RL16(src + 0);
|
|
c->status[1].step_index = src[2];
|
|
src += 4;
|
|
}
|
|
|
|
while (src < buf + buf_size) {
|
|
|
|
if (st) {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] >> 4 , 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
|
|
src[0] & 0x0F, 3);
|
|
} else {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] & 0x0F, 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] >> 4 , 3);
|
|
}
|
|
|
|
src++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WS:
|
|
/* no per-block initialization; just start decoding the data */
|
|
while (src < buf + buf_size) {
|
|
|
|
if (st) {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] >> 4 , 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[1],
|
|
src[0] & 0x0F, 3);
|
|
} else {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] >> 4 , 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
src[0] & 0x0F, 3);
|
|
}
|
|
|
|
src++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_XA:
|
|
while (buf_size >= 128) {
|
|
xa_decode(samples, src, &c->status[0], &c->status[1],
|
|
avctx->channels);
|
|
src += 128;
|
|
samples += 28 * 8;
|
|
buf_size -= 128;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_EA_EACS:
|
|
samples_in_chunk = bytestream_get_le32(&src) >> (1-st);
|
|
|
|
if (samples_in_chunk > buf_size-4-(8<<st)) {
|
|
src += buf_size - 4;
|
|
break;
|
|
}
|
|
|
|
for (i=0; i<=st; i++)
|
|
c->status[i].step_index = bytestream_get_le32(&src);
|
|
for (i=0; i<=st; i++)
|
|
c->status[i].predictor = bytestream_get_le32(&src);
|
|
|
|
for (; samples_in_chunk; samples_in_chunk--, src++) {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0], *src>>4, 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[st], *src&0x0F, 3);
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_EA_SEAD:
|
|
for (; src < buf+buf_size; src++) {
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[0] >> 4, 6);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[st],src[0]&0x0F, 6);
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_EA:
|
|
if (buf_size < 4 || AV_RL32(src) >= ((buf_size - 12) * 2)) {
|
|
src += buf_size;
|
|
break;
|
|
}
|
|
samples_in_chunk = AV_RL32(src);
|
|
src += 4;
|
|
current_left_sample = (int16_t)bytestream_get_le16(&src);
|
|
previous_left_sample = (int16_t)bytestream_get_le16(&src);
|
|
current_right_sample = (int16_t)bytestream_get_le16(&src);
|
|
previous_right_sample = (int16_t)bytestream_get_le16(&src);
|
|
|
|
for (count1 = 0; count1 < samples_in_chunk/28;count1++) {
|
|
coeff1l = ea_adpcm_table[ *src >> 4 ];
|
|
coeff2l = ea_adpcm_table[(*src >> 4 ) + 4];
|
|
coeff1r = ea_adpcm_table[*src & 0x0F];
|
|
coeff2r = ea_adpcm_table[(*src & 0x0F) + 4];
|
|
src++;
|
|
|
|
shift_left = (*src >> 4 ) + 8;
|
|
shift_right = (*src & 0x0F) + 8;
|
|
src++;
|
|
|
|
for (count2 = 0; count2 < 28; count2++) {
|
|
next_left_sample = (int32_t)((*src & 0xF0) << 24) >> shift_left;
|
|
next_right_sample = (int32_t)((*src & 0x0F) << 28) >> shift_right;
|
|
src++;
|
|
|
|
next_left_sample = (next_left_sample +
|
|
(current_left_sample * coeff1l) +
|
|
(previous_left_sample * coeff2l) + 0x80) >> 8;
|
|
next_right_sample = (next_right_sample +
|
|
(current_right_sample * coeff1r) +
|
|
(previous_right_sample * coeff2r) + 0x80) >> 8;
|
|
|
|
previous_left_sample = current_left_sample;
|
|
current_left_sample = av_clip_int16(next_left_sample);
|
|
previous_right_sample = current_right_sample;
|
|
current_right_sample = av_clip_int16(next_right_sample);
|
|
*samples++ = (unsigned short)current_left_sample;
|
|
*samples++ = (unsigned short)current_right_sample;
|
|
}
|
|
}
|
|
|
|
if (src - buf == buf_size - 2)
|
|
src += 2; // Skip terminating 0x0000
|
|
|
|
break;
|
|
case CODEC_ID_ADPCM_EA_MAXIS_XA:
|
|
for(channel = 0; channel < avctx->channels; channel++) {
|
|
for (i=0; i<2; i++)
|
|
coeff[channel][i] = ea_adpcm_table[(*src >> 4) + 4*i];
|
|
shift[channel] = (*src & 0x0F) + 8;
|
|
src++;
|
|
}
|
|
for (count1 = 0; count1 < (buf_size - avctx->channels) / avctx->channels; count1++) {
|
|
for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
|
|
for(channel = 0; channel < avctx->channels; channel++) {
|
|
int32_t sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel];
|
|
sample = (sample +
|
|
c->status[channel].sample1 * coeff[channel][0] +
|
|
c->status[channel].sample2 * coeff[channel][1] + 0x80) >> 8;
|
|
c->status[channel].sample2 = c->status[channel].sample1;
|
|
c->status[channel].sample1 = av_clip_int16(sample);
|
|
*samples++ = c->status[channel].sample1;
|
|
}
|
|
}
|
|
src+=avctx->channels;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_EA_R1:
|
|
case CODEC_ID_ADPCM_EA_R2:
|
|
case CODEC_ID_ADPCM_EA_R3: {
|
|
/* channel numbering
|
|
2chan: 0=fl, 1=fr
|
|
4chan: 0=fl, 1=rl, 2=fr, 3=rr
|
|
6chan: 0=fl, 1=c, 2=fr, 3=rl, 4=rr, 5=sub */
|
|
const int big_endian = avctx->codec->id == CODEC_ID_ADPCM_EA_R3;
|
|
int32_t previous_sample, current_sample, next_sample;
|
|
int32_t coeff1, coeff2;
|
|
uint8_t shift;
|
|
unsigned int channel;
|
|
uint16_t *samplesC;
|
|
const uint8_t *srcC;
|
|
const uint8_t *src_end = buf + buf_size;
|
|
|
|
samples_in_chunk = (big_endian ? bytestream_get_be32(&src)
|
|
: bytestream_get_le32(&src)) / 28;
|
|
if (samples_in_chunk > UINT32_MAX/(28*avctx->channels) ||
|
|
28*samples_in_chunk*avctx->channels > samples_end-samples) {
|
|
src += buf_size - 4;
|
|
break;
|
|
}
|
|
|
|
for (channel=0; channel<avctx->channels; channel++) {
|
|
int32_t offset = (big_endian ? bytestream_get_be32(&src)
|
|
: bytestream_get_le32(&src))
|
|
+ (avctx->channels-channel-1) * 4;
|
|
|
|
if ((offset < 0) || (offset >= src_end - src - 4)) break;
|
|
srcC = src + offset;
|
|
samplesC = samples + channel;
|
|
|
|
if (avctx->codec->id == CODEC_ID_ADPCM_EA_R1) {
|
|
current_sample = (int16_t)bytestream_get_le16(&srcC);
|
|
previous_sample = (int16_t)bytestream_get_le16(&srcC);
|
|
} else {
|
|
current_sample = c->status[channel].predictor;
|
|
previous_sample = c->status[channel].prev_sample;
|
|
}
|
|
|
|
for (count1=0; count1<samples_in_chunk; count1++) {
|
|
if (*srcC == 0xEE) { /* only seen in R2 and R3 */
|
|
srcC++;
|
|
if (srcC > src_end - 30*2) break;
|
|
current_sample = (int16_t)bytestream_get_be16(&srcC);
|
|
previous_sample = (int16_t)bytestream_get_be16(&srcC);
|
|
|
|
for (count2=0; count2<28; count2++) {
|
|
*samplesC = (int16_t)bytestream_get_be16(&srcC);
|
|
samplesC += avctx->channels;
|
|
}
|
|
} else {
|
|
coeff1 = ea_adpcm_table[ *srcC>>4 ];
|
|
coeff2 = ea_adpcm_table[(*srcC>>4) + 4];
|
|
shift = (*srcC++ & 0x0F) + 8;
|
|
|
|
if (srcC > src_end - 14) break;
|
|
for (count2=0; count2<28; count2++) {
|
|
if (count2 & 1)
|
|
next_sample = (int32_t)((*srcC++ & 0x0F) << 28) >> shift;
|
|
else
|
|
next_sample = (int32_t)((*srcC & 0xF0) << 24) >> shift;
|
|
|
|
next_sample += (current_sample * coeff1) +
|
|
(previous_sample * coeff2);
|
|
next_sample = av_clip_int16(next_sample >> 8);
|
|
|
|
previous_sample = current_sample;
|
|
current_sample = next_sample;
|
|
*samplesC = current_sample;
|
|
samplesC += avctx->channels;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) {
|
|
c->status[channel].predictor = current_sample;
|
|
c->status[channel].prev_sample = previous_sample;
|
|
}
|
|
}
|
|
|
|
src = src + buf_size - (4 + 4*avctx->channels);
|
|
samples += 28 * samples_in_chunk * avctx->channels;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_EA_XAS:
|
|
if (samples_end-samples < 32*4*avctx->channels
|
|
|| buf_size < (4+15)*4*avctx->channels) {
|
|
src += buf_size;
|
|
break;
|
|
}
|
|
for (channel=0; channel<avctx->channels; channel++) {
|
|
int coeff[2][4], shift[4];
|
|
short *s2, *s = &samples[channel];
|
|
for (n=0; n<4; n++, s+=32*avctx->channels) {
|
|
for (i=0; i<2; i++)
|
|
coeff[i][n] = ea_adpcm_table[(src[0]&0x0F)+4*i];
|
|
shift[n] = (src[2]&0x0F) + 8;
|
|
for (s2=s, i=0; i<2; i++, src+=2, s2+=avctx->channels)
|
|
s2[0] = (src[0]&0xF0) + (src[1]<<8);
|
|
}
|
|
|
|
for (m=2; m<32; m+=2) {
|
|
s = &samples[m*avctx->channels + channel];
|
|
for (n=0; n<4; n++, src++, s+=32*avctx->channels) {
|
|
for (s2=s, i=0; i<8; i+=4, s2+=avctx->channels) {
|
|
int level = (int32_t)((*src & (0xF0>>i)) << (24+i)) >> shift[n];
|
|
int pred = s2[-1*avctx->channels] * coeff[0][n]
|
|
+ s2[-2*avctx->channels] * coeff[1][n];
|
|
s2[0] = av_clip_int16((level + pred + 0x80) >> 8);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
samples += 32*4*avctx->channels;
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_AMV:
|
|
case CODEC_ID_ADPCM_IMA_SMJPEG:
|
|
c->status[0].predictor = (int16_t)bytestream_get_le16(&src);
|
|
c->status[0].step_index = bytestream_get_le16(&src);
|
|
|
|
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
|
|
src+=4;
|
|
|
|
while (src < buf + buf_size) {
|
|
char hi, lo;
|
|
lo = *src & 0x0F;
|
|
hi = *src >> 4;
|
|
|
|
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
|
|
FFSWAP(char, hi, lo);
|
|
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
lo, 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
|
hi, 3);
|
|
src++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_CT:
|
|
while (src < buf + buf_size) {
|
|
if (st) {
|
|
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
|
|
src[0] >> 4);
|
|
*samples++ = adpcm_ct_expand_nibble(&c->status[1],
|
|
src[0] & 0x0F);
|
|
} else {
|
|
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
|
|
src[0] >> 4);
|
|
*samples++ = adpcm_ct_expand_nibble(&c->status[0],
|
|
src[0] & 0x0F);
|
|
}
|
|
src++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_SBPRO_4:
|
|
case CODEC_ID_ADPCM_SBPRO_3:
|
|
case CODEC_ID_ADPCM_SBPRO_2:
|
|
if (!c->status[0].step_index) {
|
|
/* the first byte is a raw sample */
|
|
*samples++ = 128 * (*src++ - 0x80);
|
|
if (st)
|
|
*samples++ = 128 * (*src++ - 0x80);
|
|
c->status[0].step_index = 1;
|
|
}
|
|
if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) {
|
|
while (src < buf + buf_size) {
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
|
src[0] >> 4, 4, 0);
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
|
|
src[0] & 0x0F, 4, 0);
|
|
src++;
|
|
}
|
|
} else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) {
|
|
while (src < buf + buf_size && samples + 2 < samples_end) {
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
|
src[0] >> 5 , 3, 0);
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
|
(src[0] >> 2) & 0x07, 3, 0);
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
|
src[0] & 0x03, 2, 0);
|
|
src++;
|
|
}
|
|
} else {
|
|
while (src < buf + buf_size && samples + 3 < samples_end) {
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
|
src[0] >> 6 , 2, 2);
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
|
|
(src[0] >> 4) & 0x03, 2, 2);
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
|
(src[0] >> 2) & 0x03, 2, 2);
|
|
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
|
|
src[0] & 0x03, 2, 2);
|
|
src++;
|
|
}
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_SWF:
|
|
{
|
|
GetBitContext gb;
|
|
const int *table;
|
|
int k0, signmask, nb_bits, count;
|
|
int size = buf_size*8;
|
|
|
|
init_get_bits(&gb, buf, size);
|
|
|
|
//read bits & initial values
|
|
nb_bits = get_bits(&gb, 2)+2;
|
|
//av_log(NULL,AV_LOG_INFO,"nb_bits: %d\n", nb_bits);
|
|
table = swf_index_tables[nb_bits-2];
|
|
k0 = 1 << (nb_bits-2);
|
|
signmask = 1 << (nb_bits-1);
|
|
|
|
while (get_bits_count(&gb) <= size - 22*avctx->channels) {
|
|
for (i = 0; i < avctx->channels; i++) {
|
|
*samples++ = c->status[i].predictor = get_sbits(&gb, 16);
|
|
c->status[i].step_index = get_bits(&gb, 6);
|
|
}
|
|
|
|
for (count = 0; get_bits_count(&gb) <= size - nb_bits*avctx->channels && count < 4095; count++) {
|
|
int i;
|
|
|
|
for (i = 0; i < avctx->channels; i++) {
|
|
// similar to IMA adpcm
|
|
int delta = get_bits(&gb, nb_bits);
|
|
int step = step_table[c->status[i].step_index];
|
|
long vpdiff = 0; // vpdiff = (delta+0.5)*step/4
|
|
int k = k0;
|
|
|
|
do {
|
|
if (delta & k)
|
|
vpdiff += step;
|
|
step >>= 1;
|
|
k >>= 1;
|
|
} while(k);
|
|
vpdiff += step;
|
|
|
|
if (delta & signmask)
|
|
c->status[i].predictor -= vpdiff;
|
|
else
|
|
c->status[i].predictor += vpdiff;
|
|
|
|
c->status[i].step_index += table[delta & (~signmask)];
|
|
|
|
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
|
|
c->status[i].predictor = av_clip_int16(c->status[i].predictor);
|
|
|
|
*samples++ = c->status[i].predictor;
|
|
if (samples >= samples_end) {
|
|
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
src += buf_size;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_YAMAHA:
|
|
while (src < buf + buf_size) {
|
|
if (st) {
|
|
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
|
|
src[0] & 0x0F);
|
|
*samples++ = adpcm_yamaha_expand_nibble(&c->status[1],
|
|
src[0] >> 4 );
|
|
} else {
|
|
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
|
|
src[0] & 0x0F);
|
|
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0],
|
|
src[0] >> 4 );
|
|
}
|
|
src++;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_THP:
|
|
{
|
|
int table[2][16];
|
|
unsigned int samplecnt;
|
|
int prev[2][2];
|
|
int ch;
|
|
|
|
if (buf_size < 80) {
|
|
av_log(avctx, AV_LOG_ERROR, "frame too small\n");
|
|
return -1;
|
|
}
|
|
|
|
src+=4;
|
|
samplecnt = bytestream_get_be32(&src);
|
|
|
|
for (i = 0; i < 32; i++)
|
|
table[0][i] = (int16_t)bytestream_get_be16(&src);
|
|
|
|
/* Initialize the previous sample. */
|
|
for (i = 0; i < 4; i++)
|
|
prev[0][i] = (int16_t)bytestream_get_be16(&src);
|
|
|
|
if (samplecnt >= (samples_end - samples) / (st + 1)) {
|
|
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
|
|
return -1;
|
|
}
|
|
|
|
for (ch = 0; ch <= st; ch++) {
|
|
samples = (unsigned short *) data + ch;
|
|
|
|
/* Read in every sample for this channel. */
|
|
for (i = 0; i < samplecnt / 14; i++) {
|
|
int index = (*src >> 4) & 7;
|
|
unsigned int exp = 28 - (*src++ & 15);
|
|
int factor1 = table[ch][index * 2];
|
|
int factor2 = table[ch][index * 2 + 1];
|
|
|
|
/* Decode 14 samples. */
|
|
for (n = 0; n < 14; n++) {
|
|
int32_t sampledat;
|
|
if(n&1) sampledat= *src++ <<28;
|
|
else sampledat= (*src&0xF0)<<24;
|
|
|
|
sampledat = ((prev[ch][0]*factor1
|
|
+ prev[ch][1]*factor2) >> 11) + (sampledat>>exp);
|
|
*samples = av_clip_int16(sampledat);
|
|
prev[ch][1] = prev[ch][0];
|
|
prev[ch][0] = *samples++;
|
|
|
|
/* In case of stereo, skip one sample, this sample
|
|
is for the other channel. */
|
|
samples += st;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* In the previous loop, in case stereo is used, samples is
|
|
increased exactly one time too often. */
|
|
samples -= st;
|
|
break;
|
|
}
|
|
|
|
default:
|
|
return -1;
|
|
}
|
|
*data_size = (uint8_t *)samples - (uint8_t *)data;
|
|
return src - buf;
|
|
}
|
|
|
|
|
|
|
|
#if CONFIG_ENCODERS
|
|
#define ADPCM_ENCODER(id,name,long_name_) \
|
|
AVCodec ff_ ## name ## _encoder = { \
|
|
#name, \
|
|
AVMEDIA_TYPE_AUDIO, \
|
|
id, \
|
|
sizeof(ADPCMContext), \
|
|
adpcm_encode_init, \
|
|
adpcm_encode_frame, \
|
|
adpcm_encode_close, \
|
|
NULL, \
|
|
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \
|
|
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
|
|
}
|
|
#else
|
|
#define ADPCM_ENCODER(id,name,long_name_)
|
|
#endif
|
|
|
|
#if CONFIG_DECODERS
|
|
#define ADPCM_DECODER(id,name,long_name_) \
|
|
AVCodec ff_ ## name ## _decoder = { \
|
|
#name, \
|
|
AVMEDIA_TYPE_AUDIO, \
|
|
id, \
|
|
sizeof(ADPCMContext), \
|
|
adpcm_decode_init, \
|
|
NULL, \
|
|
NULL, \
|
|
adpcm_decode_frame, \
|
|
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
|
|
}
|
|
#else
|
|
#define ADPCM_DECODER(id,name,long_name_)
|
|
#endif
|
|
|
|
#define ADPCM_CODEC(id,name,long_name_) \
|
|
ADPCM_ENCODER(id,name,long_name_); ADPCM_DECODER(id,name,long_name_)
|
|
|
|
/* Note: Do not forget to add new entries to the Makefile as well. */
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm, "ADPCM 4X Movie");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_CT, adpcm_ct, "ADPCM Creative Technology");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_EA, adpcm_ea, "ADPCM Electronic Arts");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_EA_MAXIS_XA, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1, "ADPCM Electronic Arts R1");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
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ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
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ADPCM_DECODER(CODEC_ID_ADPCM_IMA_ISS, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
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ADPCM_CODEC (CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
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ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
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ADPCM_CODEC (CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
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ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood");
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ADPCM_CODEC (CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
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ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
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ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
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ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
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ADPCM_CODEC (CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
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ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP");
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ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA");
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ADPCM_CODEC (CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
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