Mercurial > libavcodec.hg
view adpcm.c @ 5901:35c91d1b84c0 libavcodec
grammar
| author | michael |
|---|---|
| date | Fri, 09 Nov 2007 22:15:46 +0000 |
| parents | aea495619138 |
| children | ddb692c6b01d |
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/* * 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, 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; 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: c->status[0].sample1 = c->status[0].sample2 = c->status[1].sample1 = c->status[1].sample2 = 0; 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; 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);