e4de71677f
* qatar/master: aac_latm: reconfigure decoder on audio specific config changes latmdec: fix audio specific config parsing Add avcodec_decode_audio4(). avcodec: change number of plane pointers from 4 to 8 at next major bump. Update developers documentation with coding conventions. svq1dec: avoid undefined get_bits(0) call ARM: h264dsp_neon cosmetics ARM: make some NEON macros reusable Do not memcpy raw video frames when using null muxer fate: update asf seektest vp8: flush buffers on size changes. doc: improve general documentation for MacOSX asf: use packet dts as approximation of pts asf: do not call av_read_frame rtsp: Initialize the media_type_mask in the rtp guessing demuxer Cleaned up alacenc.c Conflicts: doc/APIchanges doc/developer.texi libavcodec/8svx.c libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/nellymoserdec.c libavcodec/tta.c libavcodec/utils.c libavcodec/version.h libavcodec/wmadec.c libavformat/asfdec.c tests/ref/seek/lavf_asf Merged-by: Michael Niedermayer <michaelni@gmx.at>
1231 lines
44 KiB
C
1231 lines
44 KiB
C
/*
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* Copyright (c) 2001-2003 The ffmpeg Project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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#include "adpcm.h"
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#include "adpcm_data.h"
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/**
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* @file
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* ADPCM decoders
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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://wiki.multimedia.cx/index.php?title=Category:ADPCM_Audio_Codecs
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* http://www.pcisys.net/~melanson/codecs/simpleaudio.html [dead]
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* http://www.geocities.com/SiliconValley/8682/aud3.txt [dead]
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* http://openquicktime.sourceforge.net/
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* XAnim sources (xa_codec.c) http://xanim.polter.net/
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* http://www.cs.ucla.edu/~leec/mediabench/applications.html [dead]
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* SoX source code http://sox.sourceforge.net/
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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 [dead]
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* vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html [dead]
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* readstr http://www.geocities.co.jp/Playtown/2004/
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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,
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0, 0, -208, -220,
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0, 1, 3, 4,
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7, 8, 10, 11,
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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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/* end of tables */
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typedef struct ADPCMDecodeContext {
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AVFrame frame;
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ADPCMChannelStatus status[6];
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} ADPCMDecodeContext;
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static av_cold int adpcm_decode_init(AVCodecContext * avctx)
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{
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ADPCMDecodeContext *c = avctx->priv_data;
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unsigned int max_channels = 2;
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switch(avctx->codec->id) {
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case CODEC_ID_ADPCM_EA_R1:
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case CODEC_ID_ADPCM_EA_R2:
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case CODEC_ID_ADPCM_EA_R3:
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case CODEC_ID_ADPCM_EA_XAS:
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max_channels = 6;
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break;
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}
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if(avctx->channels > max_channels){
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return -1;
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}
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switch(avctx->codec->id) {
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case CODEC_ID_ADPCM_CT:
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c->status[0].step = c->status[1].step = 511;
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break;
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case CODEC_ID_ADPCM_IMA_WAV:
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if (avctx->bits_per_coded_sample != 4) {
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av_log(avctx, AV_LOG_ERROR, "Only 4-bit ADPCM IMA WAV files are supported\n");
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return -1;
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}
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break;
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case CODEC_ID_ADPCM_IMA_WS:
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if (avctx->extradata && avctx->extradata_size == 2 * 4) {
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c->status[0].predictor = AV_RL32(avctx->extradata);
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c->status[1].predictor = AV_RL32(avctx->extradata + 4);
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}
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break;
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default:
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break;
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avcodec_get_frame_defaults(&c->frame);
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avctx->coded_frame = &c->frame;
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return 0;
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}
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static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble, int shift)
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{
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int step_index;
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int predictor;
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int sign, delta, diff, step;
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step = ff_adpcm_step_table[c->step_index];
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step_index = c->step_index + ff_adpcm_index_table[(unsigned)nibble];
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if (step_index < 0) step_index = 0;
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else if (step_index > 88) step_index = 88;
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sign = nibble & 8;
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delta = nibble & 7;
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/* perform direct multiplication instead of series of jumps proposed by
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* the reference ADPCM implementation since modern CPUs can do the mults
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* quickly enough */
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diff = ((2 * delta + 1) * step) >> shift;
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predictor = c->predictor;
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if (sign) predictor -= diff;
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else predictor += diff;
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c->predictor = av_clip_int16(predictor);
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c->step_index = step_index;
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return (short)c->predictor;
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}
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static inline int adpcm_ima_qt_expand_nibble(ADPCMChannelStatus *c, int nibble, int shift)
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{
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int step_index;
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int predictor;
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int diff, step;
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step = ff_adpcm_step_table[c->step_index];
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step_index = c->step_index + ff_adpcm_index_table[nibble];
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step_index = av_clip(step_index, 0, 88);
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diff = step >> 3;
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if (nibble & 4) diff += step;
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if (nibble & 2) diff += step >> 1;
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if (nibble & 1) diff += step >> 2;
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if (nibble & 8)
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predictor = c->predictor - diff;
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else
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predictor = c->predictor + diff;
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c->predictor = av_clip_int16(predictor);
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c->step_index = step_index;
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return c->predictor;
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}
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static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
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{
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int predictor;
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predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
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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 = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
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if (c->idelta < 16) c->idelta = 16;
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return c->sample1;
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}
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static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
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{
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int sign, delta, diff;
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int new_step;
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sign = nibble & 8;
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delta = nibble & 7;
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/* perform direct multiplication instead of series of jumps proposed by
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* the reference ADPCM implementation since modern CPUs can do the mults
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* quickly enough */
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diff = ((2 * delta + 1) * c->step) >> 3;
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/* predictor update is not so trivial: predictor is multiplied on 254/256 before updating */
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c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff);
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c->predictor = av_clip_int16(c->predictor);
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/* calculate new step and clamp it to range 511..32767 */
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new_step = (ff_adpcm_AdaptationTable[nibble & 7] * c->step) >> 8;
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c->step = av_clip(new_step, 511, 32767);
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return (short)c->predictor;
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}
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static inline short adpcm_sbpro_expand_nibble(ADPCMChannelStatus *c, char nibble, int size, int shift)
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{
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int sign, delta, diff;
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sign = nibble & (1<<(size-1));
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delta = nibble & ((1<<(size-1))-1);
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diff = delta << (7 + c->step + shift);
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/* clamp result */
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c->predictor = av_clip(c->predictor + (sign ? -diff : diff), -16384,16256);
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/* calculate new step */
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if (delta >= (2*size - 3) && c->step < 3)
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c->step++;
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else if (delta == 0 && c->step > 0)
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c->step--;
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return (short) c->predictor;
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}
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static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned char nibble)
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{
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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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c->predictor += (c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8;
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c->predictor = av_clip_int16(c->predictor);
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c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
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c->step = av_clip(c->step, 127, 24567);
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return c->predictor;
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}
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static void xa_decode(short *out, const unsigned char *in,
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ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
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{
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int i, j;
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int shift,filter,f0,f1;
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int s_1,s_2;
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int d,s,t;
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for(i=0;i<4;i++) {
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shift = 12 - (in[4+i*2] & 15);
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filter = in[4+i*2] >> 4;
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f0 = xa_adpcm_table[filter][0];
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f1 = xa_adpcm_table[filter][1];
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s_1 = left->sample1;
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s_2 = left->sample2;
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for(j=0;j<28;j++) {
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d = in[16+i+j*4];
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t = (signed char)(d<<4)>>4;
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s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
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s_2 = s_1;
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s_1 = av_clip_int16(s);
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*out = s_1;
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out += inc;
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}
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if (inc==2) { /* stereo */
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left->sample1 = s_1;
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left->sample2 = s_2;
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s_1 = right->sample1;
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s_2 = right->sample2;
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out = out + 1 - 28*2;
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}
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shift = 12 - (in[5+i*2] & 15);
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filter = in[5+i*2] >> 4;
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f0 = xa_adpcm_table[filter][0];
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f1 = xa_adpcm_table[filter][1];
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for(j=0;j<28;j++) {
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d = in[16+i+j*4];
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t = (signed char)d >> 4;
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s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
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s_2 = s_1;
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s_1 = av_clip_int16(s);
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*out = s_1;
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out += inc;
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}
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if (inc==2) { /* stereo */
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right->sample1 = s_1;
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right->sample2 = s_2;
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out -= 1;
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} else {
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left->sample1 = s_1;
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left->sample2 = s_2;
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}
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}
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}
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/**
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* Get the number of samples that will be decoded from the packet.
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* In one case, this is actually the maximum number of samples possible to
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* decode with the given buf_size.
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*
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* @param[out] coded_samples set to the number of samples as coded in the
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* packet, or 0 if the codec does not encode the
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* number of samples in each frame.
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*/
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static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
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int buf_size, int *coded_samples)
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{
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ADPCMDecodeContext *s = avctx->priv_data;
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int nb_samples = 0;
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int ch = avctx->channels;
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int has_coded_samples = 0;
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int header_size;
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*coded_samples = 0;
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switch (avctx->codec->id) {
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/* constant, only check buf_size */
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case CODEC_ID_ADPCM_EA_XAS:
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if (buf_size < 76 * ch)
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return 0;
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nb_samples = 128;
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break;
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case CODEC_ID_ADPCM_IMA_QT:
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if (buf_size < 34 * ch)
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return 0;
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nb_samples = 64;
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break;
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/* simple 4-bit adpcm */
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case CODEC_ID_ADPCM_CT:
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case CODEC_ID_ADPCM_IMA_EA_SEAD:
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case CODEC_ID_ADPCM_IMA_WS:
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case CODEC_ID_ADPCM_YAMAHA:
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nb_samples = buf_size * 2 / ch;
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break;
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}
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if (nb_samples)
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return nb_samples;
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/* simple 4-bit adpcm, with header */
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header_size = 0;
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_4XM:
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case CODEC_ID_ADPCM_IMA_ISS: header_size = 4 * ch; break;
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case CODEC_ID_ADPCM_IMA_AMV: header_size = 8; break;
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case CODEC_ID_ADPCM_IMA_SMJPEG: header_size = 4; break;
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}
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if (header_size > 0)
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return (buf_size - header_size) * 2 / ch;
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/* more complex formats */
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_EA:
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has_coded_samples = 1;
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if (buf_size < 4)
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return 0;
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*coded_samples = AV_RL32(buf);
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*coded_samples -= *coded_samples % 28;
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nb_samples = (buf_size - 12) / 30 * 28;
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break;
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case CODEC_ID_ADPCM_IMA_EA_EACS:
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has_coded_samples = 1;
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if (buf_size < 4)
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return 0;
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*coded_samples = AV_RL32(buf);
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nb_samples = (buf_size - (4 + 8 * ch)) * 2 / ch;
|
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break;
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case CODEC_ID_ADPCM_EA_MAXIS_XA:
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nb_samples = ((buf_size - ch) / (2 * ch)) * 2 * ch;
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break;
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case CODEC_ID_ADPCM_EA_R1:
|
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case CODEC_ID_ADPCM_EA_R2:
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case CODEC_ID_ADPCM_EA_R3:
|
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/* maximum number of samples */
|
|
/* has internal offsets and a per-frame switch to signal raw 16-bit */
|
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has_coded_samples = 1;
|
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if (buf_size < 4)
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return 0;
|
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_EA_R1:
|
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header_size = 4 + 9 * ch;
|
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*coded_samples = AV_RL32(buf);
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break;
|
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case CODEC_ID_ADPCM_EA_R2:
|
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header_size = 4 + 5 * ch;
|
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*coded_samples = AV_RL32(buf);
|
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break;
|
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case CODEC_ID_ADPCM_EA_R3:
|
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header_size = 4 + 5 * ch;
|
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*coded_samples = AV_RB32(buf);
|
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break;
|
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}
|
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*coded_samples -= *coded_samples % 28;
|
|
nb_samples = (buf_size - header_size) * 2 / ch;
|
|
nb_samples -= nb_samples % 28;
|
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break;
|
|
case CODEC_ID_ADPCM_IMA_DK3:
|
|
if (avctx->block_align > 0)
|
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buf_size = FFMIN(buf_size, avctx->block_align);
|
|
nb_samples = ((buf_size - 16) * 8 / 3) / ch;
|
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break;
|
|
case CODEC_ID_ADPCM_IMA_DK4:
|
|
nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch;
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WAV:
|
|
if (avctx->block_align > 0)
|
|
buf_size = FFMIN(buf_size, avctx->block_align);
|
|
nb_samples = 1 + (buf_size - 4 * ch) / (4 * ch) * 8;
|
|
break;
|
|
case CODEC_ID_ADPCM_MS:
|
|
if (avctx->block_align > 0)
|
|
buf_size = FFMIN(buf_size, avctx->block_align);
|
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nb_samples = 2 + (buf_size - 7 * ch) * 2 / ch;
|
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break;
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case CODEC_ID_ADPCM_SBPRO_2:
|
|
case CODEC_ID_ADPCM_SBPRO_3:
|
|
case CODEC_ID_ADPCM_SBPRO_4:
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|
{
|
|
int samples_per_byte;
|
|
switch (avctx->codec->id) {
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|
case CODEC_ID_ADPCM_SBPRO_2: samples_per_byte = 4; break;
|
|
case CODEC_ID_ADPCM_SBPRO_3: samples_per_byte = 3; break;
|
|
case CODEC_ID_ADPCM_SBPRO_4: samples_per_byte = 2; break;
|
|
}
|
|
if (!s->status[0].step_index) {
|
|
nb_samples++;
|
|
buf_size -= ch;
|
|
}
|
|
nb_samples += buf_size * samples_per_byte / ch;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_SWF:
|
|
{
|
|
int buf_bits = buf_size * 8 - 2;
|
|
int nbits = (buf[0] >> 6) + 2;
|
|
int block_hdr_size = 22 * ch;
|
|
int block_size = block_hdr_size + nbits * ch * 4095;
|
|
int nblocks = buf_bits / block_size;
|
|
int bits_left = buf_bits - nblocks * block_size;
|
|
nb_samples = nblocks * 4096;
|
|
if (bits_left >= block_hdr_size)
|
|
nb_samples += 1 + (bits_left - block_hdr_size) / (nbits * ch);
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_THP:
|
|
has_coded_samples = 1;
|
|
if (buf_size < 8)
|
|
return 0;
|
|
*coded_samples = AV_RB32(&buf[4]);
|
|
*coded_samples -= *coded_samples % 14;
|
|
nb_samples = (buf_size - 80) / (8 * ch) * 14;
|
|
break;
|
|
case CODEC_ID_ADPCM_XA:
|
|
nb_samples = (buf_size / 128) * 224 / ch;
|
|
break;
|
|
}
|
|
|
|
/* validate coded sample count */
|
|
if (has_coded_samples && (*coded_samples <= 0 || *coded_samples > nb_samples))
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
return nb_samples;
|
|
}
|
|
|
|
/* DK3 ADPCM support macro */
|
|
#define DK3_GET_NEXT_NIBBLE() \
|
|
if (decode_top_nibble_next) \
|
|
{ \
|
|
nibble = last_byte >> 4; \
|
|
decode_top_nibble_next = 0; \
|
|
} \
|
|
else \
|
|
{ \
|
|
if (end_of_packet) \
|
|
break; \
|
|
last_byte = *src++; \
|
|
if (src >= buf + buf_size) \
|
|
end_of_packet = 1; \
|
|
nibble = last_byte & 0x0F; \
|
|
decode_top_nibble_next = 1; \
|
|
}
|
|
|
|
static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
ADPCMDecodeContext *c = avctx->priv_data;
|
|
ADPCMChannelStatus *cs;
|
|
int n, m, channel, i;
|
|
short *samples;
|
|
const uint8_t *src;
|
|
int st; /* stereo */
|
|
int count1, count2;
|
|
int nb_samples, coded_samples, ret;
|
|
|
|
nb_samples = get_nb_samples(avctx, buf, buf_size, &coded_samples);
|
|
if (nb_samples <= 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
/* get output buffer */
|
|
c->frame.nb_samples = nb_samples;
|
|
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
return ret;
|
|
}
|
|
samples = (short *)c->frame.data[0];
|
|
|
|
/* use coded_samples when applicable */
|
|
/* it is always <= nb_samples, so the output buffer will be large enough */
|
|
if (coded_samples) {
|
|
if (coded_samples != nb_samples)
|
|
av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n");
|
|
c->frame.nb_samples = nb_samples = coded_samples;
|
|
}
|
|
|
|
src = buf;
|
|
|
|
st = avctx->channels == 2 ? 1 : 0;
|
|
|
|
switch(avctx->codec->id) {
|
|
case CODEC_ID_ADPCM_IMA_QT:
|
|
/* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples).
|
|
Channel data is interleaved per-chunk. */
|
|
for (channel = 0; channel < avctx->channels; channel++) {
|
|
int16_t predictor;
|
|
int step_index;
|
|
cs = &(c->status[channel]);
|
|
/* (pppppp) (piiiiiii) */
|
|
|
|
/* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
|
|
predictor = AV_RB16(src);
|
|
step_index = predictor & 0x7F;
|
|
predictor &= 0xFF80;
|
|
|
|
src += 2;
|
|
|
|
if (cs->step_index == step_index) {
|
|
int diff = (int)predictor - cs->predictor;
|
|
if (diff < 0)
|
|
diff = - diff;
|
|
if (diff > 0x7f)
|
|
goto update;
|
|
} else {
|
|
update:
|
|
cs->step_index = step_index;
|
|
cs->predictor = predictor;
|
|
}
|
|
|
|
if (cs->step_index > 88){
|
|
av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index);
|
|
cs->step_index = 88;
|
|
}
|
|
|
|
samples = (short *)c->frame.data[0] + channel;
|
|
|
|
for (m = 0; m < 32; m++) {
|
|
*samples = adpcm_ima_qt_expand_nibble(cs, src[0] & 0x0F, 3);
|
|
samples += avctx->channels;
|
|
*samples = adpcm_ima_qt_expand_nibble(cs, src[0] >> 4 , 3);
|
|
samples += avctx->channels;
|
|
src ++;
|
|
}
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WAV:
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
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 */
|
|
}
|
|
|
|
for (n = (nb_samples - 1) / 8; n > 0; n--) {
|
|
for (i = 0; i < avctx->channels; i++) {
|
|
cs = &c->status[i];
|
|
for (m = 0; m < 4; m++) {
|
|
uint8_t v = *src++;
|
|
*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
|
|
samples += avctx->channels;
|
|
*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 3);
|
|
samples += avctx->channels;
|
|
}
|
|
samples -= 8 * avctx->channels - 1;
|
|
}
|
|
samples += 7 * avctx->channels;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_4XM:
|
|
for (i = 0; i < avctx->channels; i++)
|
|
c->status[i].predictor= (int16_t)bytestream_get_le16(&src);
|
|
|
|
for (i = 0; i < avctx->channels; i++) {
|
|
c->status[i].step_index= (int16_t)bytestream_get_le16(&src);
|
|
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
|
|
}
|
|
|
|
for (i = 0; i < avctx->channels; i++) {
|
|
samples = (short *)c->frame.data[0] + i;
|
|
cs = &c->status[i];
|
|
for (n = nb_samples >> 1; n > 0; n--, src++) {
|
|
uint8_t v = *src;
|
|
*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
|
|
samples += avctx->channels;
|
|
*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 4);
|
|
samples += avctx->channels;
|
|
}
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_MS:
|
|
{
|
|
int block_predictor;
|
|
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
block_predictor = av_clip(*src++, 0, 6);
|
|
c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor];
|
|
c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor];
|
|
if (st) {
|
|
block_predictor = av_clip(*src++, 0, 6);
|
|
c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor];
|
|
c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor];
|
|
}
|
|
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].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 = (nb_samples - 2) >> (1 - st); n > 0; n--, src++) {
|
|
*samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
|
|
*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
|
|
}
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_IMA_DK4:
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
for (channel = 0; channel < avctx->channels; channel++) {
|
|
cs = &c->status[channel];
|
|
cs->predictor = (int16_t)bytestream_get_le16(&src);
|
|
cs->step_index = *src++;
|
|
src++;
|
|
*samples++ = cs->predictor;
|
|
}
|
|
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
|
uint8_t v = *src;
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_DK3:
|
|
{
|
|
unsigned char last_byte = 0;
|
|
unsigned char nibble;
|
|
int decode_top_nibble_next = 0;
|
|
int end_of_packet = 0;
|
|
int diff_channel;
|
|
|
|
if (avctx->block_align != 0 && buf_size > avctx->block_align)
|
|
buf_size = avctx->block_align;
|
|
|
|
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:
|
|
for (channel = 0; channel < avctx->channels; channel++) {
|
|
cs = &c->status[channel];
|
|
cs->predictor = (int16_t)bytestream_get_le16(&src);
|
|
cs->step_index = *src++;
|
|
src++;
|
|
}
|
|
|
|
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
|
uint8_t v1, v2;
|
|
uint8_t v = *src;
|
|
/* nibbles are swapped for mono */
|
|
if (st) {
|
|
v1 = v >> 4;
|
|
v2 = v & 0x0F;
|
|
} else {
|
|
v2 = v >> 4;
|
|
v1 = v & 0x0F;
|
|
}
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v1, 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3);
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_WS:
|
|
while (src < buf + buf_size) {
|
|
uint8_t v = *src++;
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[0], v >> 4 , 3);
|
|
*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
|
|
}
|
|
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:
|
|
src += 4; // skip sample count (already read)
|
|
|
|
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 (n = nb_samples >> (1 - st); n > 0; n--, 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 (n = nb_samples >> (1 - st); n > 0; n--, 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:
|
|
{
|
|
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;
|
|
|
|
/* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces,
|
|
each coding 28 stereo samples. */
|
|
|
|
src += 4; // skip sample count (already read)
|
|
|
|
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 < nb_samples / 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 = 20 - (*src >> 4);
|
|
shift_right = 20 - (*src & 0x0F);
|
|
src++;
|
|
|
|
for (count2 = 0; count2 < 28; count2++) {
|
|
next_left_sample = sign_extend(*src >> 4, 4) << shift_left;
|
|
next_right_sample = sign_extend(*src, 4) << 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:
|
|
{
|
|
int coeff[2][2], shift[2];
|
|
|
|
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] = 20 - (*src & 0x0F);
|
|
src++;
|
|
}
|
|
for (count1 = 0; count1 < nb_samples / 2; 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 = sign_extend(src[channel] >> i, 4) << 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;
|
|
}
|
|
/* consume whole packet */
|
|
src = buf + buf_size;
|
|
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;
|
|
int count = 0;
|
|
|
|
src += 4; // skip sample count (already read)
|
|
|
|
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 < nb_samples / 28; 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 = 20 - (*srcC++ & 0x0F);
|
|
|
|
if (srcC > src_end - 14) break;
|
|
for (count2=0; count2<28; count2++) {
|
|
if (count2 & 1)
|
|
next_sample = sign_extend(*srcC++, 4) << shift;
|
|
else
|
|
next_sample = sign_extend(*srcC >> 4, 4) << 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 (!count) {
|
|
count = count1;
|
|
} else if (count != count1) {
|
|
av_log(avctx, AV_LOG_WARNING, "per-channel sample count mismatch\n");
|
|
count = FFMAX(count, count1);
|
|
}
|
|
|
|
if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) {
|
|
c->status[channel].predictor = current_sample;
|
|
c->status[channel].prev_sample = previous_sample;
|
|
}
|
|
}
|
|
|
|
c->frame.nb_samples = count * 28;
|
|
src = src_end;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_EA_XAS:
|
|
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] = 20 - (src[2] & 0x0F);
|
|
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 = sign_extend(*src >> (4 - i), 4) << 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);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
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;
|
|
|
|
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
|
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);
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_CT:
|
|
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
|
uint8_t v = *src;
|
|
*samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4 );
|
|
*samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F);
|
|
}
|
|
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;
|
|
nb_samples--;
|
|
}
|
|
if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) {
|
|
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
|
*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);
|
|
}
|
|
} else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) {
|
|
for (n = nb_samples / 3; n > 0; n--, src++) {
|
|
*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);
|
|
}
|
|
} else {
|
|
for (n = nb_samples >> (2 - st); n > 0; n--, src++) {
|
|
*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);
|
|
}
|
|
}
|
|
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 = ff_adpcm_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;
|
|
}
|
|
}
|
|
}
|
|
src += buf_size;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_YAMAHA:
|
|
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
|
uint8_t v = *src;
|
|
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F);
|
|
*samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4 );
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_THP:
|
|
{
|
|
int table[2][16];
|
|
int prev[2][2];
|
|
int ch;
|
|
|
|
src += 4; // skip channel size
|
|
src += 4; // skip number of samples (already read)
|
|
|
|
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);
|
|
|
|
for (ch = 0; ch <= st; ch++) {
|
|
samples = (short *)c->frame.data[0] + ch;
|
|
|
|
/* Read in every sample for this channel. */
|
|
for (i = 0; i < nb_samples / 14; i++) {
|
|
int index = (*src >> 4) & 7;
|
|
unsigned int exp = *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 = sign_extend(*src++, 4);
|
|
else sampledat = sign_extend(*src >> 4, 4);
|
|
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
default:
|
|
return -1;
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
*(AVFrame *)data = c->frame;
|
|
|
|
return src - buf;
|
|
}
|
|
|
|
|
|
#define ADPCM_DECODER(id_, name_, long_name_) \
|
|
AVCodec ff_ ## name_ ## _decoder = { \
|
|
.name = #name_, \
|
|
.type = AVMEDIA_TYPE_AUDIO, \
|
|
.id = id_, \
|
|
.priv_data_size = sizeof(ADPCMDecodeContext), \
|
|
.init = adpcm_decode_init, \
|
|
.decode = adpcm_decode_frame, \
|
|
.capabilities = CODEC_CAP_DR1, \
|
|
.long_name = NULL_IF_CONFIG_SMALL(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");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_ISS, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA");
|
|
ADPCM_DECODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
|