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Diffstat (limited to 'contrib/ffmpeg/libavcodec/adpcm.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/adpcm.c | 1602 |
1 files changed, 1602 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/adpcm.c b/contrib/ffmpeg/libavcodec/adpcm.c new file mode 100644 index 000000000..eadcfaedd --- /dev/null +++ b/contrib/ffmpeg/libavcodec/adpcm.c @@ -0,0 +1,1602 @@ +/* + * ADPCM codecs + * Copyright (c) 2001-2003 The ffmpeg Project + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ +#include "avcodec.h" +#include "bitstream.h" +#include "bytestream.h" + +/** + * @file adpcm.c + * ADPCM codecs. + * First version by Francois Revol (revol@free.fr) + * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood) + * by Mike Melanson (melanson@pcisys.net) + * CD-ROM XA ADPCM codec by BERO + * EA ADPCM decoder by Robin Kay (komadori@myrealbox.com) + * EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org) + * EA IMA EACS decoder by Peter Ross (pross@xvid.org) + * EA IMA SEAD decoder by Peter Ross (pross@xvid.org) + * EA ADPCM XAS decoder by Peter Ross (pross@xvid.org) + * THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl) + * + * Features and limitations: + * + * Reference documents: + * http://www.pcisys.net/~melanson/codecs/simpleaudio.html + * http://www.geocities.com/SiliconValley/8682/aud3.txt + * http://openquicktime.sourceforge.net/plugins.htm + * XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html + * http://www.cs.ucla.edu/~leec/mediabench/applications.html + * SoX source code http://home.sprynet.com/~cbagwell/sox.html + * + * CD-ROM XA: + * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html + * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html + * readstr http://www.geocities.co.jp/Playtown/2004/ + */ + +#define BLKSIZE 1024 + +/* step_table[] and index_table[] are from the ADPCM reference source */ +/* This is the index table: */ +static const int index_table[16] = { + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8, +}; + +/** + * This is the step table. Note that many programs use slight deviations from + * this table, but such deviations are negligible: + */ +static const int step_table[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 +}; + +/* These are for MS-ADPCM */ +/* AdaptationTable[], AdaptCoeff1[], and AdaptCoeff2[] are from libsndfile */ +static const int AdaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 +}; + +static const int AdaptCoeff1[] = { + 256, 512, 0, 192, 240, 460, 392 +}; + +static const int AdaptCoeff2[] = { + 0, -256, 0, 64, 0, -208, -232 +}; + +/* These are for CD-ROM XA ADPCM */ +static const int xa_adpcm_table[5][2] = { + { 0, 0 }, + { 60, 0 }, + { 115, -52 }, + { 98, -55 }, + { 122, -60 } +}; + +static const int ea_adpcm_table[] = { + 0, 240, 460, 392, 0, 0, -208, -220, 0, 1, + 3, 4, 7, 8, 10, 11, 0, -1, -3, -4 +}; + +static const int ct_adpcm_table[8] = { + 0x00E6, 0x00E6, 0x00E6, 0x00E6, + 0x0133, 0x0199, 0x0200, 0x0266 +}; + +// padded to zero where table size is less then 16 +static const int swf_index_tables[4][16] = { + /*2*/ { -1, 2 }, + /*3*/ { -1, -1, 2, 4 }, + /*4*/ { -1, -1, -1, -1, 2, 4, 6, 8 }, + /*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 } +}; + +static const int yamaha_indexscale[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 230, 230, 230, 230, 307, 409, 512, 614 +}; + +static const int yamaha_difflookup[] = { + 1, 3, 5, 7, 9, 11, 13, 15, + -1, -3, -5, -7, -9, -11, -13, -15 +}; + +/* end of tables */ + +typedef struct ADPCMChannelStatus { + int predictor; + short int step_index; + int step; + /* for encoding */ + int prev_sample; + + /* MS version */ + short sample1; + short sample2; + int coeff1; + int coeff2; + int idelta; +} ADPCMChannelStatus; + +typedef struct ADPCMContext { + int channel; /* for stereo MOVs, decode left, then decode right, then tell it's decoded */ + ADPCMChannelStatus status[6]; +} ADPCMContext; + +/* XXX: implement encoding */ + +#ifdef CONFIG_ENCODERS +static int adpcm_encode_init(AVCodecContext *avctx) +{ + if (avctx->channels > 2) + return -1; /* only stereo or mono =) */ + switch(avctx->codec->id) { + case CODEC_ID_ADPCM_IMA_WAV: + avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 / (4 * avctx->channels) + 1; /* each 16 bits sample gives one nibble */ + /* and we have 4 bytes per channel overhead */ + avctx->block_align = BLKSIZE; + /* seems frame_size isn't taken into account... have to buffer the samples :-( */ + break; + case CODEC_ID_ADPCM_MS: + avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */ + /* and we have 7 bytes per channel overhead */ + avctx->block_align = BLKSIZE; + break; + case CODEC_ID_ADPCM_YAMAHA: + avctx->frame_size = BLKSIZE * avctx->channels; + avctx->block_align = BLKSIZE; + break; + case CODEC_ID_ADPCM_SWF: + if (avctx->sample_rate != 11025 && + avctx->sample_rate != 22050 && + avctx->sample_rate != 44100) { + av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, 22050 or 44100\n"); + return -1; + } + avctx->frame_size = 512 * (avctx->sample_rate / 11025); + break; + default: + return -1; + break; + } + + avctx->coded_frame= avcodec_alloc_frame(); + avctx->coded_frame->key_frame= 1; + + return 0; +} + +static int adpcm_encode_close(AVCodecContext *avctx) +{ + av_freep(&avctx->coded_frame); + + return 0; +} + + +static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, short sample) +{ + int delta = sample - c->prev_sample; + int nibble = FFMIN(7, abs(delta)*4/step_table[c->step_index]) + (delta<0)*8; + c->prev_sample += ((step_table[c->step_index] * yamaha_difflookup[nibble]) / 8); + c->prev_sample = av_clip_int16(c->prev_sample); + c->step_index = av_clip(c->step_index + index_table[nibble], 0, 88); + return nibble; +} + +static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, short sample) +{ + int predictor, nibble, bias; + + predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256; + + nibble= sample - predictor; + if(nibble>=0) bias= c->idelta/2; + else bias=-c->idelta/2; + + nibble= (nibble + bias) / c->idelta; + nibble= av_clip(nibble, -8, 7)&0x0F; + + 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 nibble; +} + +static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, short sample) +{ + int nibble, delta; + + if(!c->step) { + c->predictor = 0; + c->step = 127; + } + + delta = sample - c->predictor; + + nibble = FFMIN(7, abs(delta)*4/c->step) + (delta<0)*8; + + 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 nibble; +} + +typedef struct TrellisPath { + int nibble; + int prev; +} TrellisPath; + +typedef struct TrellisNode { + uint32_t ssd; + int path; + int sample1; + int sample2; + int step; +} TrellisNode; + +static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples, + uint8_t *dst, ADPCMChannelStatus *c, int n) +{ +#define FREEZE_INTERVAL 128 + //FIXME 6% faster if frontier is a compile-time constant + const int frontier = 1 << avctx->trellis; + const int stride = avctx->channels; + const int version = avctx->codec->id; + const int max_paths = frontier*FREEZE_INTERVAL; + TrellisPath paths[max_paths], *p; + TrellisNode node_buf[2][frontier]; + TrellisNode *nodep_buf[2][frontier]; + TrellisNode **nodes = nodep_buf[0]; // nodes[] is always sorted by .ssd + TrellisNode **nodes_next = nodep_buf[1]; + int pathn = 0, froze = -1, i, j, k; + + assert(!(max_paths&(max_paths-1))); + + memset(nodep_buf, 0, sizeof(nodep_buf)); + nodes[0] = &node_buf[1][0]; + nodes[0]->ssd = 0; + nodes[0]->path = 0; + nodes[0]->step = c->step_index; + nodes[0]->sample1 = c->sample1; + nodes[0]->sample2 = c->sample2; + if((version == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_SWF)) + nodes[0]->sample1 = c->prev_sample; + if(version == CODEC_ID_ADPCM_MS) + nodes[0]->step = c->idelta; + if(version == CODEC_ID_ADPCM_YAMAHA) { + if(c->step == 0) { + nodes[0]->step = 127; + nodes[0]->sample1 = 0; + } else { + nodes[0]->step = c->step; + nodes[0]->sample1 = c->predictor; + } + } + + for(i=0; i<n; i++) { + TrellisNode *t = node_buf[i&1]; + TrellisNode **u; + int sample = samples[i*stride]; + memset(nodes_next, 0, frontier*sizeof(TrellisNode*)); + for(j=0; j<frontier && nodes[j]; j++) { + // higher j have higher ssd already, so they're unlikely to use a suboptimal next sample too + const int range = (j < frontier/2) ? 1 : 0; + const int step = nodes[j]->step; + int nidx; + if(version == CODEC_ID_ADPCM_MS) { + const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 256; + const int div = (sample - predictor) / step; + const int nmin = av_clip(div-range, -8, 6); + const int nmax = av_clip(div+range, -7, 7); + for(nidx=nmin; nidx<=nmax; nidx++) { + const int nibble = nidx & 0xf; + int dec_sample = predictor + nidx * step; +#define STORE_NODE(NAME, STEP_INDEX)\ + int d;\ + uint32_t ssd;\ + dec_sample = av_clip_int16(dec_sample);\ + d = sample - dec_sample;\ + ssd = nodes[j]->ssd + d*d;\ + if(nodes_next[frontier-1] && ssd >= nodes_next[frontier-1]->ssd)\ + continue;\ + /* Collapse any two states with the same previous sample value. \ + * One could also distinguish states by step and by 2nd to last + * sample, but the effects of that are negligible. */\ + for(k=0; k<frontier && nodes_next[k]; k++) {\ + if(dec_sample == nodes_next[k]->sample1) {\ + assert(ssd >= nodes_next[k]->ssd);\ + goto next_##NAME;\ + }\ + }\ + for(k=0; k<frontier; k++) {\ + if(!nodes_next[k] || ssd < nodes_next[k]->ssd) {\ + TrellisNode *u = nodes_next[frontier-1];\ + if(!u) {\ + assert(pathn < max_paths);\ + u = t++;\ + u->path = pathn++;\ + }\ + u->ssd = ssd;\ + u->step = STEP_INDEX;\ + u->sample2 = nodes[j]->sample1;\ + u->sample1 = dec_sample;\ + paths[u->path].nibble = nibble;\ + paths[u->path].prev = nodes[j]->path;\ + memmove(&nodes_next[k+1], &nodes_next[k], (frontier-k-1)*sizeof(TrellisNode*));\ + nodes_next[k] = u;\ + break;\ + }\ + }\ + next_##NAME:; + STORE_NODE(ms, FFMAX(16, (AdaptationTable[nibble] * step) >> 8)); + } + } else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (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; + + // 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; + + 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) { + uint8_t buf[2][n*8]; + adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n*8); + if(avctx->channels == 2) + adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n*8); + for(i=0; i<n; i++) { + *dst++ = buf[0][8*i+0] | (buf[0][8*i+1] << 4); + *dst++ = buf[0][8*i+2] | (buf[0][8*i+3] << 4); + *dst++ = buf[0][8*i+4] | (buf[0][8*i+5] << 4); + *dst++ = buf[0][8*i+6] | (buf[0][8*i+7] << 4); + if (avctx->channels == 2) { + *dst++ = buf[1][8*i+0] | (buf[1][8*i+1] << 4); + *dst++ = buf[1][8*i+2] | (buf[1][8*i+3] << 4); + *dst++ = buf[1][8*i+4] | (buf[1][8*i+5] << 4); + *dst++ = buf[1][8*i+6] | (buf[1][8*i+7] << 4); + } + } + } 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_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_bits(&pb, 16, samples[i] & 0xFFFF); + put_bits(&pb, 6, c->status[i].step_index); + c->status[i].prev_sample = (signed short)samples[i]; + } + + if(avctx->trellis > 0) { + uint8_t buf[2][n]; + adpcm_compress_trellis(avctx, samples+2, buf[0], &c->status[0], n); + if (avctx->channels == 2) + adpcm_compress_trellis(avctx, samples+3, buf[1], &c->status[1], n); + for(i=0; i<n; i++) { + put_bits(&pb, 4, buf[0][i]); + if (avctx->channels == 2) + put_bits(&pb, 4, buf[1][i]); + } + } 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].sample1= *samples++; + + bytestream_put_le16(&dst, c->status[i].sample1); + } + for(i=0; i<avctx->channels; i++){ + c->status[i].sample2= *samples++; + + bytestream_put_le16(&dst, c->status[i].sample2); + } + + if(avctx->trellis > 0) { + int n = avctx->block_align - 7*avctx->channels; + uint8_t buf[2][n]; + if(avctx->channels == 1) { + n *= 2; + adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n); + for(i=0; i<n; i+=2) + *dst++ = (buf[0][i] << 4) | buf[0][i+1]; + } else { + adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n); + adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n); + for(i=0; i<n; i++) + *dst++ = (buf[0][i] << 4) | buf[1][i]; + } + } 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) { + uint8_t buf[2][n*2]; + n *= 2; + if(avctx->channels == 1) { + adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n); + for(i=0; i<n; i+=2) + *dst++ = buf[0][i] | (buf[0][i+1] << 4); + } else { + adpcm_compress_trellis(avctx, samples, buf[0], &c->status[0], n); + adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n); + for(i=0; i<n; i++) + *dst++ = buf[0][i] | (buf[1][i] << 4); + } + } else + for (; n>0; n--) { + for(i = 0; i < avctx->channels; i++) { + int nibble; + nibble = adpcm_yamaha_compress_sample(&c->status[i], samples[i]); + nibble |= adpcm_yamaha_compress_sample(&c->status[i], samples[i+avctx->channels]) << 4; + *dst++ = nibble; + } + samples += 2 * avctx->channels; + } + break; + default: + return -1; + } + return dst - frame; +} +#endif //CONFIG_ENCODERS + +static 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_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; + } + 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))) / 256; + 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 = (ct_adpcm_table[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) & 0x0F; \ + 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, + const uint8_t *buf, int buf_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; + + 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);/* >> 2*avctx->channels;*/ + channel = c->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]; + + if (st && channel) + samples++; + + 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) & 0x0F, 3); + samples += avctx->channels; + src ++; + } + + if(st) { /* handle stereo interlacing */ + c->channel = (channel + 1) % 2; /* we get one packet for left, then one for right data */ + if(channel == 1) { /* wait for the other packet before outputing anything */ + return src - buf; + } + } + 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)(src[0] + (src[1]<<8)); + src+=2; + + 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)(src[0] + (src[1]<<8)); src+=2; + if(st){ + c->status[1].predictor= (int16_t)(src[0] + (src[1]<<8)); src+=2; + } + c->status[0].step_index= (int16_t)(src[0] + (src[1]<<8)); src+=2; + if(st){ + c->status[1].step_index= (int16_t)(src[0] + (src[1]<<8)); src+=2; + } + 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, 7); + block_predictor[1] = 0; + if (st) + block_predictor[1] = av_clip(*src++, 0, 7); + c->status[0].idelta = (int16_t)((*src & 0xFF) | ((src[1] << 8) & 0xFF00)); + src+=2; + if (st){ + c->status[1].idelta = (int16_t)((*src & 0xFF) | ((src[1] << 8) & 0xFF00)); + src+=2; + } + 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 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00)); + src+=2; + if (st) c->status[1].sample1 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00)); + if (st) src+=2; + c->status[0].sample2 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00)); + src+=2; + if (st) c->status[1].sample2 = ((*src & 0xFF) | ((src[1] << 8) & 0xFF00)); + if (st) src+=2; + + *samples++ = c->status[0].sample1; + if (st) *samples++ = c->status[1].sample1; + *samples++ = c->status[0].sample2; + if (st) *samples++ = c->status[1].sample2; + for(;n>0;n--) { + *samples++ = adpcm_ms_expand_nibble(&c->status[0], (src[0] >> 4) & 0x0F); + *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)(src[0] | (src[1] << 8)); + c->status[0].step_index = src[2]; + src += 4; + *samples++ = c->status[0].predictor; + if (st) { + c->status[1].predictor = (int16_t)(src[0] | (src[1] << 8)); + c->status[1].step_index = src[2]; + src += 4; + *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) & 0x0F, 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)(src[10] | (src[11] << 8)); + c->status[1].predictor = (int16_t)(src[12] | (src[13] << 8)); + 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_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) & 0x0F, 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) & 0x0F, 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: + samples_in_chunk = AV_RL32(src); + if (samples_in_chunk >= ((buf_size - 12) * 2)) { + src += buf_size; + break; + } + src += 4; + current_left_sample = (int16_t)AV_RL16(src); + src += 2; + previous_left_sample = (int16_t)AV_RL16(src); + src += 2; + current_right_sample = (int16_t)AV_RL16(src); + src += 2; + previous_right_sample = (int16_t)AV_RL16(src); + src += 2; + + for (count1 = 0; count1 < samples_in_chunk/28;count1++) { + coeff1l = ea_adpcm_table[(*src >> 4) & 0x0F]; + coeff2l = ea_adpcm_table[((*src >> 4) & 0x0F) + 4]; + coeff1r = ea_adpcm_table[*src & 0x0F]; + coeff2r = ea_adpcm_table[(*src & 0x0F) + 4]; + src++; + + shift_left = ((*src >> 4) & 0x0F) + 8; + shift_right = (*src & 0x0F) + 8; + src++; + + for (count2 = 0; count2 < 28; count2++) { + next_left_sample = (((*src & 0xF0) << 24) >> shift_left); + next_right_sample = (((*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; + } + } + 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; + + 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++) { + srcC = src + (big_endian ? bytestream_get_be32(&src) + : bytestream_get_le32(&src)) + + (avctx->channels-channel-1) * 4; + 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++; + 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) & 0x0F ]; + coeff2 = ea_adpcm_table[((*srcC>>4) & 0x0F) + 4]; + shift = (*srcC++ & 0x0F) + 8; + + for (count2=0; count2<28; count2++) { + if (count2 & 1) + next_sample = ((*srcC++ & 0x0F) << 28) >> shift; + else + next_sample = ((*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 = ((*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) & 0x0F; + + 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) & 0x0F); + *samples++ = adpcm_ct_expand_nibble(&c->status[1], + src[0] & 0x0F); + } else { + *samples++ = adpcm_ct_expand_nibble(&c->status[0], + (src[0] >> 4) & 0x0F); + *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) & 0x0F, 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) & 0x07, 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) & 0x03, 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) & 0x0F); + } else { + *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], + src[0] & 0x0F); + *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], + (src[0] >> 4) & 0x0F); + } + 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; +} + + + +#ifdef CONFIG_ENCODERS +#define ADPCM_ENCODER(id,name) \ +AVCodec name ## _encoder = { \ + #name, \ + CODEC_TYPE_AUDIO, \ + id, \ + sizeof(ADPCMContext), \ + adpcm_encode_init, \ + adpcm_encode_frame, \ + adpcm_encode_close, \ + NULL, \ +}; +#else +#define ADPCM_ENCODER(id,name) +#endif + +#ifdef CONFIG_DECODERS +#define ADPCM_DECODER(id,name) \ +AVCodec name ## _decoder = { \ + #name, \ + CODEC_TYPE_AUDIO, \ + id, \ + sizeof(ADPCMContext), \ + adpcm_decode_init, \ + NULL, \ + NULL, \ + adpcm_decode_frame, \ +}; +#else +#define ADPCM_DECODER(id,name) +#endif + +#define ADPCM_CODEC(id, name) \ +ADPCM_ENCODER(id,name) ADPCM_DECODER(id,name) + +ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm); +ADPCM_DECODER(CODEC_ID_ADPCM_CT, adpcm_ct); +ADPCM_DECODER(CODEC_ID_ADPCM_EA, adpcm_ea); +ADPCM_DECODER(CODEC_ID_ADPCM_EA_R1, adpcm_ea_r1); +ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2); +ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3); +ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg); +ADPCM_CODEC (CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws); +ADPCM_CODEC (CODEC_ID_ADPCM_MS, adpcm_ms); +ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4); +ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3); +ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2); +ADPCM_CODEC (CODEC_ID_ADPCM_SWF, adpcm_swf); +ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp); +ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa); +ADPCM_CODEC (CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha); |