X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fadpcm.c;h=b319635ed441fd09ed1b36bdf4c8f78d60b01a78;hb=3ab770001817e0f52114a9876819f07fcd8ed93a;hp=7ddd72375a9fe929deb01c4d5fcddaac41aa6742;hpb=7a00bbad2100367481240e62876b941b5c4befdc;p=ffmpeg diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c index 7ddd72375a9..b319635ed44 100644 --- a/libavcodec/adpcm.c +++ b/libavcodec/adpcm.c @@ -1,30 +1,32 @@ /* - * ADPCM codecs * Copyright (c) 2001-2003 The ffmpeg Project * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; 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 "get_bits.h" +#include "put_bits.h" #include "bytestream.h" +#include "adpcm.h" +#include "adpcm_data.h" /** - * @file libavcodec/adpcm.c - * ADPCM codecs. + * @file + * ADPCM decoders * First version by Francois Revol (revol@free.fr) * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood) * by Mike Melanson (melanson@pcisys.net) @@ -40,71 +42,35 @@ * 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 + * http://wiki.multimedia.cx/index.php?title=Category:ADPCM_Audio_Codecs + * http://www.pcisys.net/~melanson/codecs/simpleaudio.html [dead] + * http://www.geocities.com/SiliconValley/8682/aud3.txt [dead] + * http://openquicktime.sourceforge.net/ + * XAnim sources (xa_codec.c) http://xanim.polter.net/ + * http://www.cs.ucla.edu/~leec/mediabench/applications.html [dead] + * SoX source code http://sox.sourceforge.net/ * * CD-ROM XA: - * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html - * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html + * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html [dead] + * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html [dead] * 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 uint8_t AdaptCoeff1[] = { - 64, 128, 0, 48, 60, 115, 98 -}; - -static const int8_t AdaptCoeff2[] = { - 0, -64, 0, 16, 0, -52, -58 -}; - /* 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 } + { 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 + 0, 240, 460, 392, + 0, 0, -208, -220, + 0, 1, 3, 4, + 7, 8, 10, 11, + 0, -1, -3, -4 }; // padded to zero where table size is less then 16 @@ -115,575 +81,45 @@ static const int swf_index_tables[4][16] = { /*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 { +typedef struct ADPCMDecodeContext { + AVFrame frame; ADPCMChannelStatus status[6]; -} ADPCMContext; - -/* XXX: implement encoding */ - -#if CONFIG_ENCODERS -static av_cold int adpcm_encode_init(AVCodecContext *avctx) -{ - if (avctx->channels > 2) - return -1; /* only stereo or mono =) */ - - if(avctx->trellis && (unsigned)avctx->trellis > 16U){ - av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n"); - return -1; - } - - 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_IMA_QT: - avctx->frame_size = 64; - avctx->block_align = 34 * avctx->channels; - 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 av_cold 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))) / 64; - - 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_IMA_QT) || (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; istep; - int nidx; - if(version == CODEC_ID_ADPCM_MS) { - const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 64; - 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; ksample1) {\ - assert(ssd >= nodes_next[k]->ssd);\ - goto next_##NAME;\ - }\ - }\ - for(k=0; kssd) {\ - 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_IMA_QT)|| (version == CODEC_ID_ADPCM_SWF)) { -#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\ - const int predictor = nodes[j]->sample1;\ - const int div = (sample - predictor) * 4 / STEP_TABLE;\ - int nmin = av_clip(div-range, -7, 6);\ - int nmax = av_clip(div+range, -6, 7);\ - if(nmin<=0) nmin--; /* distinguish -0 from +0 */\ - if(nmax<0) nmax--;\ - for(nidx=nmin; nidx<=nmax; nidx++) {\ - const int nibble = nidx<0 ? 7-nidx : nidx;\ - int dec_sample = predictor + (STEP_TABLE * yamaha_difflookup[nibble]) / 8;\ - STORE_NODE(NAME, STEP_INDEX);\ - } - LOOP_NODES(ima, step_table[step], av_clip(step + index_table[nibble], 0, 88)); - } else { //CODEC_ID_ADPCM_YAMAHA - LOOP_NODES(yamaha, step, av_clip((step * yamaha_indexscale[nibble]) >> 8, 127, 24567)); -#undef LOOP_NODES -#undef STORE_NODE - } - } - - u = nodes; - nodes = nodes_next; - nodes_next = u; - - // prevent overflow - if(nodes[0]->ssd > (1<<28)) { - for(j=1; jssd -= 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; ichannels == 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_IMA_QT: - { - int ch, i; - PutBitContext pb; - init_put_bits(&pb, dst, buf_size*8); - - for(ch=0; chchannels; ch++){ - put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7); - put_bits(&pb, 7, c->status[ch].step_index); - if(avctx->trellis > 0) { - uint8_t buf[64]; - adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64); - for(i=0; i<64; i++) - put_bits(&pb, 4, buf[i^1]); - c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F; - } else { - for (i=0; i<64; i+=2){ - int t1, t2; - t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]); - t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]); - put_bits(&pb, 4, t2); - put_bits(&pb, 4, t1); - } - c->status[ch].prev_sample &= ~0x7F; - } - } - - dst += put_bits_count(&pb)>>3; - break; - } - case CODEC_ID_ADPCM_SWF: - { - int i; - PutBitContext pb; - init_put_bits(&pb, dst, buf_size*8); - - n = avctx->frame_size-1; - - //Store AdpcmCodeSize - put_bits(&pb, 2, 2); //Set 4bits flash adpcm format - - //Init the encoder state - for(i=0; ichannels; i++){ - c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits - put_sbits(&pb, 16, samples[i]); - put_bits(&pb, 6, c->status[i].step_index); - c->status[i].prev_sample = (signed short)samples[i]; - } - - if(avctx->trellis > 0) { - 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; ichannels == 2) - put_bits(&pb, 4, buf[1][i]); - } - } else { - for (i=1; iframe_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; ichannels; i++){ - int predictor=0; - - *dst++ = predictor; - c->status[i].coeff1 = AdaptCoeff1[predictor]; - c->status[i].coeff2 = AdaptCoeff2[predictor]; - } - for(i=0; ichannels; i++){ - if (c->status[i].idelta < 16) - c->status[i].idelta = 16; - - bytestream_put_le16(&dst, c->status[i].idelta); - } - for(i=0; ichannels; i++){ - c->status[i].sample2= *samples++; - } - for(i=0; ichannels; i++){ - c->status[i].sample1= *samples++; - - bytestream_put_le16(&dst, c->status[i].sample1); - } - for(i=0; ichannels; i++) - 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; istatus[0], n); - adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n); - for(i=0; ichannels; iblock_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; istatus[0], n); - adpcm_compress_trellis(avctx, samples+1, buf[1], &c->status[1], n); - for(i=0; i0; 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 +} ADPCMDecodeContext; static av_cold int adpcm_decode_init(AVCodecContext * avctx) { - ADPCMContext *c = avctx->priv_data; + ADPCMDecodeContext *c = avctx->priv_data; + unsigned int min_channels = 1; unsigned int max_channels = 2; switch(avctx->codec->id) { + case CODEC_ID_ADPCM_EA: + min_channels = 2; + break; case CODEC_ID_ADPCM_EA_R1: case CODEC_ID_ADPCM_EA_R2: case CODEC_ID_ADPCM_EA_R3: + case CODEC_ID_ADPCM_EA_XAS: max_channels = 6; break; } - if(avctx->channels > max_channels){ - return -1; + if (avctx->channels < min_channels || avctx->channels > max_channels) { + av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); + return AVERROR(EINVAL); } switch(avctx->codec->id) { case CODEC_ID_ADPCM_CT: c->status[0].step = c->status[1].step = 511; break; + case CODEC_ID_ADPCM_IMA_WAV: + if (avctx->bits_per_coded_sample != 4) { + av_log(avctx, AV_LOG_ERROR, "Only 4-bit ADPCM IMA WAV files are supported\n"); + return -1; + } + break; case CODEC_ID_ADPCM_IMA_WS: if (avctx->extradata && avctx->extradata_size == 2 * 4) { c->status[0].predictor = AV_RL32(avctx->extradata); @@ -693,7 +129,11 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx) default: break; } - avctx->sample_fmt = SAMPLE_FMT_S16; + avctx->sample_fmt = AV_SAMPLE_FMT_S16; + + avcodec_get_frame_defaults(&c->frame); + avctx->coded_frame = &c->frame; + return 0; } @@ -703,8 +143,8 @@ static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble, int predictor; int sign, delta, diff, step; - step = step_table[c->step_index]; - step_index = c->step_index + index_table[(unsigned)nibble]; + step = ff_adpcm_step_table[c->step_index]; + step_index = c->step_index + ff_adpcm_index_table[(unsigned)nibble]; if (step_index < 0) step_index = 0; else if (step_index > 88) step_index = 88; @@ -724,6 +164,32 @@ static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble, return (short)c->predictor; } +static inline int adpcm_ima_qt_expand_nibble(ADPCMChannelStatus *c, int nibble, int shift) +{ + int step_index; + int predictor; + int diff, step; + + step = ff_adpcm_step_table[c->step_index]; + step_index = c->step_index + ff_adpcm_index_table[nibble]; + step_index = av_clip(step_index, 0, 88); + + diff = step >> 3; + if (nibble & 4) diff += step; + if (nibble & 2) diff += step >> 1; + if (nibble & 1) diff += step >> 2; + + if (nibble & 8) + predictor = c->predictor - diff; + else + predictor = c->predictor + diff; + + c->predictor = av_clip_int16(predictor); + c->step_index = step_index; + + return c->predictor; +} + static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble) { int predictor; @@ -733,7 +199,7 @@ static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble) c->sample2 = c->sample1; c->sample1 = av_clip_int16(predictor); - c->idelta = (AdaptationTable[(int)nibble] * c->idelta) >> 8; + c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8; if (c->idelta < 16) c->idelta = 16; return c->sample1; @@ -754,7 +220,7 @@ static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble) c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff); c->predictor = av_clip_int16(c->predictor); /* calculate new step and clamp it to range 511..32767 */ - new_step = (AdaptationTable[nibble & 7] * c->step) >> 8; + new_step = (ff_adpcm_AdaptationTable[nibble & 7] * c->step) >> 8; c->step = av_clip(new_step, 511, 32767); return (short)c->predictor; @@ -787,9 +253,9 @@ static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned c c->step = 127; } - c->predictor += (c->step * yamaha_difflookup[nibble]) / 8; + c->predictor += (c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8; c->predictor = av_clip_int16(c->predictor); - c->step = (c->step * yamaha_indexscale[nibble]) >> 8; + c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8; c->step = av_clip(c->step, 127, 24567); return c->predictor; } @@ -859,6 +325,173 @@ static void xa_decode(short *out, const unsigned char *in, } } +/** + * Get the number of samples that will be decoded from the packet. + * In one case, this is actually the maximum number of samples possible to + * decode with the given buf_size. + * + * @param[out] coded_samples set to the number of samples as coded in the + * packet, or 0 if the codec does not encode the + * number of samples in each frame. + */ +static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf, + int buf_size, int *coded_samples) +{ + ADPCMDecodeContext *s = avctx->priv_data; + int nb_samples = 0; + int ch = avctx->channels; + int has_coded_samples = 0; + int header_size; + + *coded_samples = 0; + + switch (avctx->codec->id) { + /* constant, only check buf_size */ + case CODEC_ID_ADPCM_EA_XAS: + if (buf_size < 76 * ch) + return 0; + nb_samples = 128; + break; + case CODEC_ID_ADPCM_IMA_QT: + if (buf_size < 34 * ch) + return 0; + nb_samples = 64; + break; + /* simple 4-bit adpcm */ + case CODEC_ID_ADPCM_CT: + case CODEC_ID_ADPCM_IMA_EA_SEAD: + case CODEC_ID_ADPCM_IMA_WS: + case CODEC_ID_ADPCM_YAMAHA: + nb_samples = buf_size * 2 / ch; + break; + } + if (nb_samples) + return nb_samples; + + /* simple 4-bit adpcm, with header */ + header_size = 0; + switch (avctx->codec->id) { + case CODEC_ID_ADPCM_4XM: + case CODEC_ID_ADPCM_IMA_ISS: header_size = 4 * ch; break; + case CODEC_ID_ADPCM_IMA_AMV: header_size = 8; break; + case CODEC_ID_ADPCM_IMA_SMJPEG: header_size = 4; break; + } + if (header_size > 0) + return (buf_size - header_size) * 2 / ch; + + /* more complex formats */ + switch (avctx->codec->id) { + case CODEC_ID_ADPCM_EA: + has_coded_samples = 1; + if (buf_size < 4) + return 0; + *coded_samples = AV_RL32(buf); + *coded_samples -= *coded_samples % 28; + nb_samples = (buf_size - 12) / 30 * 28; + break; + case CODEC_ID_ADPCM_IMA_EA_EACS: + has_coded_samples = 1; + if (buf_size < 4) + return 0; + *coded_samples = AV_RL32(buf); + nb_samples = (buf_size - (4 + 8 * ch)) * 2 / ch; + break; + case CODEC_ID_ADPCM_EA_MAXIS_XA: + nb_samples = ((buf_size - ch) / (2 * ch)) * 2 * ch; + break; + case CODEC_ID_ADPCM_EA_R1: + case CODEC_ID_ADPCM_EA_R2: + case CODEC_ID_ADPCM_EA_R3: + /* maximum number of samples */ + /* has internal offsets and a per-frame switch to signal raw 16-bit */ + has_coded_samples = 1; + if (buf_size < 4) + return 0; + switch (avctx->codec->id) { + case CODEC_ID_ADPCM_EA_R1: + header_size = 4 + 9 * ch; + *coded_samples = AV_RL32(buf); + break; + case CODEC_ID_ADPCM_EA_R2: + header_size = 4 + 5 * ch; + *coded_samples = AV_RL32(buf); + break; + case CODEC_ID_ADPCM_EA_R3: + header_size = 4 + 5 * ch; + *coded_samples = AV_RB32(buf); + break; + } + *coded_samples -= *coded_samples % 28; + nb_samples = (buf_size - header_size) * 2 / ch; + nb_samples -= nb_samples % 28; + break; + case CODEC_ID_ADPCM_IMA_DK3: + if (avctx->block_align > 0) + buf_size = FFMIN(buf_size, avctx->block_align); + nb_samples = ((buf_size - 16) * 8 / 3) / ch; + break; + case CODEC_ID_ADPCM_IMA_DK4: + nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch; + break; + case CODEC_ID_ADPCM_IMA_WAV: + if (avctx->block_align > 0) + buf_size = FFMIN(buf_size, avctx->block_align); + nb_samples = 1 + (buf_size - 4 * ch) / (4 * ch) * 8; + break; + case CODEC_ID_ADPCM_MS: + if (avctx->block_align > 0) + buf_size = FFMIN(buf_size, avctx->block_align); + nb_samples = 2 + (buf_size - 7 * ch) * 2 / ch; + break; + case CODEC_ID_ADPCM_SBPRO_2: + case CODEC_ID_ADPCM_SBPRO_3: + case CODEC_ID_ADPCM_SBPRO_4: + { + int samples_per_byte; + switch (avctx->codec->id) { + case CODEC_ID_ADPCM_SBPRO_2: samples_per_byte = 4; break; + case CODEC_ID_ADPCM_SBPRO_3: samples_per_byte = 3; break; + case CODEC_ID_ADPCM_SBPRO_4: samples_per_byte = 2; break; + } + if (!s->status[0].step_index) { + nb_samples++; + buf_size -= ch; + } + nb_samples += buf_size * samples_per_byte / ch; + break; + } + case CODEC_ID_ADPCM_SWF: + { + int buf_bits = buf_size * 8 - 2; + int nbits = (buf[0] >> 6) + 2; + int block_hdr_size = 22 * ch; + int block_size = block_hdr_size + nbits * ch * 4095; + int nblocks = buf_bits / block_size; + int bits_left = buf_bits - nblocks * block_size; + nb_samples = nblocks * 4096; + if (bits_left >= block_hdr_size) + nb_samples += 1 + (bits_left - block_hdr_size) / (nbits * ch); + break; + } + case CODEC_ID_ADPCM_THP: + has_coded_samples = 1; + if (buf_size < 8) + return 0; + *coded_samples = AV_RB32(&buf[4]); + *coded_samples -= *coded_samples % 14; + nb_samples = (buf_size - 80) / (8 * ch) * 14; + break; + case CODEC_ID_ADPCM_XA: + nb_samples = (buf_size / 128) * 224 / ch; + break; + } + + /* validate coded sample count */ + if (has_coded_samples && (*coded_samples <= 0 || *coded_samples > nb_samples)) + return AVERROR_INVALIDDATA; + + return nb_samples; +} /* DK3 ADPCM support macro */ #define DK3_GET_NEXT_NIBBLE() \ @@ -869,105 +502,104 @@ static void xa_decode(short *out, const unsigned char *in, } \ else \ { \ + if (end_of_packet) \ + break; \ last_byte = *src++; \ - if (src >= buf + buf_size) break; \ + if (src >= buf + buf_size) \ + end_of_packet = 1; \ nibble = last_byte & 0x0F; \ decode_top_nibble_next = 1; \ } -static int adpcm_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int adpcm_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - ADPCMContext *c = avctx->priv_data; + ADPCMDecodeContext *c = avctx->priv_data; ADPCMChannelStatus *cs; int n, m, channel, i; - int block_predictor[2]; short *samples; - short *samples_end; const uint8_t *src; int st; /* stereo */ - - /* DK3 ADPCM accounting variables */ - unsigned char last_byte = 0; - unsigned char nibble; - int decode_top_nibble_next = 0; - int diff_channel; - - /* EA ADPCM state variables */ - uint32_t samples_in_chunk; - int32_t previous_left_sample, previous_right_sample; - int32_t current_left_sample, current_right_sample; - int32_t next_left_sample, next_right_sample; - int32_t coeff1l, coeff2l, coeff1r, coeff2r; - uint8_t shift_left, shift_right; int count1, count2; - int coeff[2][2], shift[2];//used in EA MAXIS ADPCM + int nb_samples, coded_samples, ret; - if (!buf_size) - return 0; + nb_samples = get_nb_samples(avctx, buf, buf_size, &coded_samples); + if (nb_samples <= 0) { + av_log(avctx, AV_LOG_ERROR, "invalid number of samples in packet\n"); + return AVERROR_INVALIDDATA; + } - //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; + /* get output buffer */ + c->frame.nb_samples = nb_samples; + if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + samples = (short *)c->frame.data[0]; + + /* use coded_samples when applicable */ + /* it is always <= nb_samples, so the output buffer will be large enough */ + if (coded_samples) { + if (coded_samples != nb_samples) + av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n"); + c->frame.nb_samples = nb_samples = coded_samples; + } - samples = data; - samples_end= samples + *data_size/2; - *data_size= 0; src = buf; st = avctx->channels == 2 ? 1 : 0; switch(avctx->codec->id) { case CODEC_ID_ADPCM_IMA_QT: - n = buf_size - 2*avctx->channels; + /* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples). + Channel data is interleaved per-chunk. */ for (channel = 0; channel < avctx->channels; channel++) { + int16_t predictor; + int step_index; cs = &(c->status[channel]); /* (pppppp) (piiiiiii) */ /* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */ - 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; + predictor = AV_RB16(src); + step_index = predictor & 0x7F; + predictor &= 0xFF80; + + src += 2; + + if (cs->step_index == step_index) { + int diff = (int)predictor - cs->predictor; + if (diff < 0) + diff = - diff; + if (diff > 0x7f) + goto update; + } else { + update: + cs->step_index = step_index; + cs->predictor = predictor; + } if (cs->step_index > 88){ av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n", cs->step_index); cs->step_index = 88; } - cs->step = step_table[cs->step_index]; + samples = (short *)c->frame.data[0] + channel; - samples = (short*)data + channel; - - for(m=32; n>0 && m>0; n--, m--) { /* in QuickTime, IMA is encoded by chuncks of 34 bytes (=64 samples) */ - *samples = adpcm_ima_expand_nibble(cs, src[0] & 0x0F, 3); + for (m = 0; m < 32; m++) { + *samples = adpcm_ima_qt_expand_nibble(cs, src[0] & 0x0F, 3); samples += avctx->channels; - *samples = adpcm_ima_expand_nibble(cs, src[0] >> 4 , 3); + *samples = adpcm_ima_qt_expand_nibble(cs, src[0] >> 4 , 3); samples += avctx->channels; src ++; } } - if (st) - samples--; break; case CODEC_ID_ADPCM_IMA_WAV: if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; -// samples_per_block= (block_align-4*chanels)*8 / (bits_per_sample * chanels) + 1; - for(i=0; ichannels; i++){ cs = &(c->status[i]); cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src); @@ -980,61 +612,61 @@ static int adpcm_decode_frame(AVCodecContext *avctx, 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++; + for (n = (nb_samples - 1) / 8; n > 0; n--) { + for (i = 0; i < avctx->channels; i++) { + cs = &c->status[i]; + for (m = 0; m < 4; m++) { + uint8_t v = *src++; + *samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 3); + samples += avctx->channels; + *samples = adpcm_ima_expand_nibble(cs, v >> 4 , 3); + samples += avctx->channels; + } + samples -= 8 * avctx->channels - 1; } - src += 4*st; + samples += 7 * avctx->channels; } break; case CODEC_ID_ADPCM_4XM: - cs = &(c->status[0]); - c->status[0].predictor= (int16_t)bytestream_get_le16(&src); - if(st){ - c->status[1].predictor= (int16_t)bytestream_get_le16(&src); - } - c->status[0].step_index= (int16_t)bytestream_get_le16(&src); - if(st){ - c->status[1].step_index= (int16_t)bytestream_get_le16(&src); - } - if (cs->step_index < 0) cs->step_index = 0; - if (cs->step_index > 88) cs->step_index = 88; + for (i = 0; i < avctx->channels; i++) + c->status[i].predictor= (int16_t)bytestream_get_le16(&src); - m= (buf_size - (src - buf))>>st; - for(i=0; istatus[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); + for (i = 0; i < avctx->channels; i++) { + c->status[i].step_index= (int16_t)bytestream_get_le16(&src); + c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88); } - src += m<channels; i++) { + samples = (short *)c->frame.data[0] + i; + cs = &c->status[i]; + for (n = nb_samples >> 1; n > 0; n--, src++) { + uint8_t v = *src; + *samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 4); + samples += avctx->channels; + *samples = adpcm_ima_expand_nibble(cs, v >> 4 , 4); + samples += avctx->channels; + } + } break; case CODEC_ID_ADPCM_MS: + { + int block_predictor; + if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; - n = buf_size - 7 * avctx->channels; - if (n < 0) - return -1; - block_predictor[0] = av_clip(*src++, 0, 6); - block_predictor[1] = 0; - if (st) - block_predictor[1] = av_clip(*src++, 0, 6); + + block_predictor = av_clip(*src++, 0, 6); + c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor]; + c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor]; + if (st) { + block_predictor = av_clip(*src++, 0, 6); + c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor]; + c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor]; + } c->status[0].idelta = (int16_t)bytestream_get_le16(&src); if (st){ c->status[1].idelta = (int16_t)bytestream_get_le16(&src); } - c->status[0].coeff1 = AdaptCoeff1[block_predictor[0]]; - c->status[0].coeff2 = AdaptCoeff2[block_predictor[0]]; - c->status[1].coeff1 = AdaptCoeff1[block_predictor[1]]; - c->status[1].coeff2 = AdaptCoeff2[block_predictor[1]]; c->status[0].sample1 = bytestream_get_le16(&src); if (st) c->status[1].sample1 = bytestream_get_le16(&src); @@ -1045,51 +677,40 @@ static int adpcm_decode_frame(AVCodecContext *avctx, if (st) *samples++ = c->status[1].sample2; *samples++ = c->status[0].sample1; if (st) *samples++ = c->status[1].sample1; - for(;n>0;n--) { + for(n = (nb_samples - 2) >> (1 - st); n > 0; n--, src++) { *samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 ); *samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F); - src ++; } break; + } case CODEC_ID_ADPCM_IMA_DK4: if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; - c->status[0].predictor = (int16_t)bytestream_get_le16(&src); - c->status[0].step_index = *src++; - src++; - *samples++ = c->status[0].predictor; - if (st) { - c->status[1].predictor = (int16_t)bytestream_get_le16(&src); - c->status[1].step_index = *src++; + for (channel = 0; channel < avctx->channels; channel++) { + cs = &c->status[channel]; + cs->predictor = (int16_t)bytestream_get_le16(&src); + cs->step_index = *src++; src++; - *samples++ = c->status[1].predictor; + *samples++ = cs->predictor; } - while (src < buf + buf_size) { - - /* take care of the top nibble (always left or mono channel) */ - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4, 3); - - /* take care of the bottom nibble, which is right sample for - * stereo, or another mono sample */ - if (st) - *samples++ = adpcm_ima_expand_nibble(&c->status[1], - src[0] & 0x0F, 3); - else - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] & 0x0F, 3); - - src++; + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { + uint8_t v = *src; + *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3); + *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3); } break; case CODEC_ID_ADPCM_IMA_DK3: + { + unsigned char last_byte = 0; + unsigned char nibble; + int decode_top_nibble_next = 0; + int end_of_packet = 0; + int diff_channel; + if (avctx->block_align != 0 && buf_size > avctx->block_align) buf_size = avctx->block_align; - if(buf_size + 16 > (samples_end - samples)*3/8) - return -1; - c->status[0].predictor = (int16_t)AV_RL16(src + 10); c->status[1].predictor = (int16_t)AV_RL16(src + 12); c->status[0].step_index = src[14]; @@ -1128,50 +749,35 @@ static int adpcm_decode_frame(AVCodecContext *avctx, *samples++ = c->status[0].predictor - c->status[1].predictor; } break; + } case CODEC_ID_ADPCM_IMA_ISS: - c->status[0].predictor = (int16_t)AV_RL16(src + 0); - c->status[0].step_index = src[2]; - src += 4; - if(st) { - c->status[1].predictor = (int16_t)AV_RL16(src + 0); - c->status[1].step_index = src[2]; - src += 4; + for (channel = 0; channel < avctx->channels; channel++) { + cs = &c->status[channel]; + cs->predictor = (int16_t)bytestream_get_le16(&src); + cs->step_index = *src++; + src++; } - while (src < buf + buf_size) { - + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { + uint8_t v1, v2; + uint8_t v = *src; + /* nibbles are swapped for mono */ if (st) { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[1], - src[0] & 0x0F, 3); + v1 = v >> 4; + v2 = v & 0x0F; } else { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] & 0x0F, 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); + v2 = v >> 4; + v1 = v & 0x0F; } - - src++; + *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v1, 3); + *samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3); } break; case CODEC_ID_ADPCM_IMA_WS: - /* no per-block initialization; just start decoding the data */ while (src < buf + buf_size) { - - if (st) { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[1], - src[0] & 0x0F, 3); - } else { - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] >> 4 , 3); - *samples++ = adpcm_ima_expand_nibble(&c->status[0], - src[0] & 0x0F, 3); - } - - src++; + uint8_t v = *src++; + *samples++ = adpcm_ima_expand_nibble(&c->status[0], v >> 4 , 3); + *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3); } break; case CODEC_ID_ADPCM_XA: @@ -1184,55 +790,56 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } 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<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++) { + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { *samples++ = adpcm_ima_expand_nibble(&c->status[0], *src>>4, 3); *samples++ = adpcm_ima_expand_nibble(&c->status[st], *src&0x0F, 3); } break; case CODEC_ID_ADPCM_IMA_EA_SEAD: - for (; src < buf+buf_size; src++) { + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { *samples++ = adpcm_ima_expand_nibble(&c->status[0], src[0] >> 4, 6); *samples++ = adpcm_ima_expand_nibble(&c->status[st],src[0]&0x0F, 6); } break; case CODEC_ID_ADPCM_EA: - samples_in_chunk = AV_RL32(src); - if (samples_in_chunk >= ((buf_size - 12) * 2)) { - src += buf_size; - break; - } - src += 4; + { + int32_t previous_left_sample, previous_right_sample; + int32_t current_left_sample, current_right_sample; + int32_t next_left_sample, next_right_sample; + int32_t coeff1l, coeff2l, coeff1r, coeff2r; + uint8_t shift_left, shift_right; + + /* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces, + each coding 28 stereo samples. */ + + src += 4; // skip sample count (already read) + current_left_sample = (int16_t)bytestream_get_le16(&src); previous_left_sample = (int16_t)bytestream_get_le16(&src); current_right_sample = (int16_t)bytestream_get_le16(&src); previous_right_sample = (int16_t)bytestream_get_le16(&src); - for (count1 = 0; count1 < samples_in_chunk/28;count1++) { + for (count1 = 0; count1 < nb_samples / 28; count1++) { coeff1l = ea_adpcm_table[ *src >> 4 ]; coeff2l = ea_adpcm_table[(*src >> 4 ) + 4]; coeff1r = ea_adpcm_table[*src & 0x0F]; coeff2r = ea_adpcm_table[(*src & 0x0F) + 4]; src++; - shift_left = (*src >> 4 ) + 8; - shift_right = (*src & 0x0F) + 8; + shift_left = 20 - (*src >> 4); + shift_right = 20 - (*src & 0x0F); src++; for (count2 = 0; count2 < 28; count2++) { - next_left_sample = (int32_t)((*src & 0xF0) << 24) >> shift_left; - next_right_sample = (int32_t)((*src & 0x0F) << 28) >> shift_right; + next_left_sample = sign_extend(*src >> 4, 4) << shift_left; + next_right_sample = sign_extend(*src, 4) << shift_right; src++; next_left_sample = (next_left_sample + @@ -1250,18 +857,26 @@ static int adpcm_decode_frame(AVCodecContext *avctx, *samples++ = (unsigned short)current_right_sample; } } + + if (src - buf == buf_size - 2) + src += 2; // Skip terminating 0x0000 + break; + } case CODEC_ID_ADPCM_EA_MAXIS_XA: + { + int coeff[2][2], shift[2]; + for(channel = 0; channel < avctx->channels; channel++) { for (i=0; i<2; i++) coeff[channel][i] = ea_adpcm_table[(*src >> 4) + 4*i]; - shift[channel] = (*src & 0x0F) + 8; + shift[channel] = 20 - (*src & 0x0F); src++; } - for (count1 = 0; count1 < (buf_size - avctx->channels) / avctx->channels; count1++) { + for (count1 = 0; count1 < nb_samples / 2; count1++) { for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */ for(channel = 0; channel < avctx->channels; channel++) { - int32_t sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel]; + int32_t sample = sign_extend(src[channel] >> i, 4) << shift[channel]; sample = (sample + c->status[channel].sample1 * coeff[channel][0] + c->status[channel].sample2 * coeff[channel][1] + 0x80) >> 8; @@ -1272,7 +887,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } src+=avctx->channels; } + /* consume whole packet */ + src = buf + buf_size; break; + } case CODEC_ID_ADPCM_EA_R1: case CODEC_ID_ADPCM_EA_R2: case CODEC_ID_ADPCM_EA_R3: { @@ -1288,14 +906,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx, uint16_t *samplesC; const uint8_t *srcC; const uint8_t *src_end = buf + buf_size; + int count = 0; - 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; - } + src += 4; // skip sample count (already read) for (channel=0; channelchannels; channel++) { int32_t offset = (big_endian ? bytestream_get_be32(&src) @@ -1314,7 +927,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, previous_sample = c->status[channel].prev_sample; } - for (count1=0; count1 src_end - 30*2) break; @@ -1328,14 +941,14 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } else { coeff1 = ea_adpcm_table[ *srcC>>4 ]; coeff2 = ea_adpcm_table[(*srcC>>4) + 4]; - shift = (*srcC++ & 0x0F) + 8; + shift = 20 - (*srcC++ & 0x0F); if (srcC > src_end - 14) break; for (count2=0; count2<28; count2++) { if (count2 & 1) - next_sample = (int32_t)((*srcC++ & 0x0F) << 28) >> shift; + next_sample = sign_extend(*srcC++, 4) << shift; else - next_sample = (int32_t)((*srcC & 0xF0) << 24) >> shift; + next_sample = sign_extend(*srcC >> 4, 4) << shift; next_sample += (current_sample * coeff1) + (previous_sample * coeff2); @@ -1348,6 +961,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } } } + if (!count) { + count = count1; + } else if (count != count1) { + av_log(avctx, AV_LOG_WARNING, "per-channel sample count mismatch\n"); + count = FFMAX(count, count1); + } if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) { c->status[channel].predictor = current_sample; @@ -1355,23 +974,18 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } } - src = src + buf_size - (4 + 4*avctx->channels); - samples += 28 * samples_in_chunk * avctx->channels; + c->frame.nb_samples = count * 28; + src = src_end; 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; channelchannels; 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; + shift[n] = 20 - (src[2] & 0x0F); for (s2=s, i=0; i<2; i++, src+=2, s2+=avctx->channels) s2[0] = (src[0]&0xF0) + (src[1]<<8); } @@ -1380,7 +994,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, s = &samples[m*avctx->channels + channel]; for (n=0; n<4; n++, src++, s+=32*avctx->channels) { for (s2=s, i=0; i<8; i+=4, s2+=avctx->channels) { - int level = (int32_t)((*src & (0xF0>>i)) << (24+i)) >> shift[n]; + int level = sign_extend(*src >> (4 - i), 4) << shift[n]; int pred = s2[-1*avctx->channels] * coeff[0][n] + s2[-2*avctx->channels] * coeff[1][n]; s2[0] = av_clip_int16((level + pred + 0x80) >> 8); @@ -1388,17 +1002,20 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } } } - 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; + if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV) { + c->status[0].predictor = sign_extend(bytestream_get_le16(&src), 16); + c->status[0].step_index = bytestream_get_le16(&src); + src += 4; + } else { + c->status[0].predictor = sign_extend(bytestream_get_be16(&src), 16); + c->status[0].step_index = bytestream_get_byte(&src); + src += 1; + } - while (src < buf + buf_size) { + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { char hi, lo; lo = *src & 0x0F; hi = *src >> 4; @@ -1410,23 +1027,13 @@ static int adpcm_decode_frame(AVCodecContext *avctx, lo, 3); *samples++ = adpcm_ima_expand_nibble(&c->status[0], hi, 3); - src++; } break; case CODEC_ID_ADPCM_CT: - while (src < buf + buf_size) { - if (st) { - *samples++ = adpcm_ct_expand_nibble(&c->status[0], - src[0] >> 4); - *samples++ = adpcm_ct_expand_nibble(&c->status[1], - src[0] & 0x0F); - } else { - *samples++ = adpcm_ct_expand_nibble(&c->status[0], - src[0] >> 4); - *samples++ = adpcm_ct_expand_nibble(&c->status[0], - src[0] & 0x0F); - } - src++; + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { + uint8_t v = *src; + *samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4 ); + *samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F); } break; case CODEC_ID_ADPCM_SBPRO_4: @@ -1438,27 +1045,26 @@ static int adpcm_decode_frame(AVCodecContext *avctx, if (st) *samples++ = 128 * (*src++ - 0x80); c->status[0].step_index = 1; + nb_samples--; } if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) { - while (src < buf + buf_size) { + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { *samples++ = adpcm_sbpro_expand_nibble(&c->status[0], src[0] >> 4, 4, 0); *samples++ = adpcm_sbpro_expand_nibble(&c->status[st], src[0] & 0x0F, 4, 0); - src++; } } else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) { - while (src < buf + buf_size && samples + 2 < samples_end) { + for (n = nb_samples / 3; n > 0; n--, src++) { *samples++ = adpcm_sbpro_expand_nibble(&c->status[0], src[0] >> 5 , 3, 0); *samples++ = adpcm_sbpro_expand_nibble(&c->status[0], (src[0] >> 2) & 0x07, 3, 0); *samples++ = adpcm_sbpro_expand_nibble(&c->status[0], src[0] & 0x03, 2, 0); - src++; } } else { - while (src < buf + buf_size && samples + 3 < samples_end) { + for (n = nb_samples >> (2 - st); n > 0; n--, src++) { *samples++ = adpcm_sbpro_expand_nibble(&c->status[0], src[0] >> 6 , 2, 2); *samples++ = adpcm_sbpro_expand_nibble(&c->status[st], @@ -1467,7 +1073,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx, (src[0] >> 2) & 0x03, 2, 2); *samples++ = adpcm_sbpro_expand_nibble(&c->status[st], src[0] & 0x03, 2, 2); - src++; } } break; @@ -1499,7 +1104,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, 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]; + int step = ff_adpcm_step_table[c->status[i].step_index]; long vpdiff = 0; // vpdiff = (delta+0.5)*step/4 int k = k0; @@ -1522,10 +1127,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx, 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; - } } } } @@ -1533,35 +1134,20 @@ static int adpcm_decode_frame(AVCodecContext *avctx, break; } case CODEC_ID_ADPCM_YAMAHA: - while (src < buf + buf_size) { - if (st) { - *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], - src[0] & 0x0F); - *samples++ = adpcm_yamaha_expand_nibble(&c->status[1], - src[0] >> 4 ); - } else { - *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], - src[0] & 0x0F); - *samples++ = adpcm_yamaha_expand_nibble(&c->status[0], - src[0] >> 4 ); - } - src++; + for (n = nb_samples >> (1 - st); n > 0; n--, src++) { + uint8_t v = *src; + *samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F); + *samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4 ); } break; case CODEC_ID_ADPCM_THP: { int table[2][16]; - 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); + src += 4; // skip channel size + src += 4; // skip number of samples (already read) for (i = 0; i < 32; i++) table[0][i] = (int16_t)bytestream_get_be16(&src); @@ -1570,29 +1156,24 @@ static int adpcm_decode_frame(AVCodecContext *avctx, 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; + samples = (short *)c->frame.data[0] + ch; /* Read in every sample for this channel. */ - for (i = 0; i < samplecnt / 14; i++) { + for (i = 0; i < nb_samples / 14; i++) { int index = (*src >> 4) & 7; - unsigned int exp = 28 - (*src++ & 15); + unsigned int exp = *src++ & 15; int factor1 = table[ch][index * 2]; int factor2 = table[ch][index * 2 + 1]; /* Decode 14 samples. */ for (n = 0; n < 14; n++) { int32_t sampledat; - if(n&1) sampledat= *src++ <<28; - else sampledat= (*src&0xF0)<<24; + if(n&1) sampledat = sign_extend(*src++, 4); + else sampledat = sign_extend(*src >> 4, 4); sampledat = ((prev[ch][0]*factor1 - + prev[ch][1]*factor2) >> 11) + (sampledat>>exp); + + prev[ch][1]*factor2) >> 11) + (sampledat << exp); *samples = av_clip_int16(sampledat); prev[ch][1] = prev[ch][0]; prev[ch][0] = *samples++; @@ -1603,59 +1184,31 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } } } - - /* 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; -} + *got_frame_ptr = 1; + *(AVFrame *)data = c->frame; + return src - buf; +} -#if CONFIG_ENCODERS -#define ADPCM_ENCODER(id,name,long_name_) \ -AVCodec name ## _encoder = { \ - #name, \ - CODEC_TYPE_AUDIO, \ - id, \ - sizeof(ADPCMContext), \ - adpcm_encode_init, \ - adpcm_encode_frame, \ - adpcm_encode_close, \ - NULL, \ - .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, \ - .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ -}; -#else -#define ADPCM_ENCODER(id,name,long_name_) -#endif - -#if CONFIG_DECODERS -#define ADPCM_DECODER(id,name,long_name_) \ -AVCodec name ## _decoder = { \ - #name, \ - CODEC_TYPE_AUDIO, \ - id, \ - sizeof(ADPCMContext), \ - adpcm_decode_init, \ - NULL, \ - NULL, \ - adpcm_decode_frame, \ - .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ -}; -#else -#define ADPCM_DECODER(id,name,long_name_) -#endif -#define ADPCM_CODEC(id,name,long_name_) \ - ADPCM_ENCODER(id,name,long_name_) ADPCM_DECODER(id,name,long_name_) +#define ADPCM_DECODER(id_, name_, long_name_) \ +AVCodec ff_ ## name_ ## _decoder = { \ + .name = #name_, \ + .type = AVMEDIA_TYPE_AUDIO, \ + .id = id_, \ + .priv_data_size = sizeof(ADPCMDecodeContext), \ + .init = adpcm_decode_init, \ + .decode = adpcm_decode_frame, \ + .capabilities = CODEC_CAP_DR1, \ + .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ +} /* Note: Do not forget to add new entries to the Makefile as well. */ ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm, "ADPCM 4X Movie"); @@ -1672,15 +1225,15 @@ ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4"); ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS"); ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_SEAD, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD"); ADPCM_DECODER(CODEC_ID_ADPCM_IMA_ISS, adpcm_ima_iss, "ADPCM IMA Funcom ISS"); -ADPCM_CODEC (CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime"); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime"); ADPCM_DECODER(CODEC_ID_ADPCM_IMA_SMJPEG, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG"); -ADPCM_CODEC (CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV"); +ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV"); ADPCM_DECODER(CODEC_ID_ADPCM_IMA_WS, adpcm_ima_ws, "ADPCM IMA Westwood"); -ADPCM_CODEC (CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft"); +ADPCM_DECODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft"); ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_2, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit"); ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_3, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit"); ADPCM_DECODER(CODEC_ID_ADPCM_SBPRO_4, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit"); -ADPCM_CODEC (CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash"); +ADPCM_DECODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash"); ADPCM_DECODER(CODEC_ID_ADPCM_THP, adpcm_thp, "ADPCM Nintendo Gamecube THP"); ADPCM_DECODER(CODEC_ID_ADPCM_XA, adpcm_xa, "ADPCM CDROM XA"); -ADPCM_CODEC (CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha"); +ADPCM_DECODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");