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author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-09-07 18:34:09 -0400 |
---|---|---|
committer | Justin Ruggles <justin.ruggles@gmail.com> | 2011-09-12 11:26:11 -0400 |
commit | 826c56d16e55f3819a75d01f957dd295aa1e9f3a (patch) | |
tree | 65ffaad2a28d590487ca681c343de2b0a79a48b6 /libavcodec | |
parent | 57650c70e22b8259f4ac65d5826a667c8f67726e (diff) | |
download | ffmpeg-streaming-826c56d16e55f3819a75d01f957dd295aa1e9f3a.zip ffmpeg-streaming-826c56d16e55f3819a75d01f957dd295aa1e9f3a.tar.gz |
adpcm: split ADPCM encoders and decoders into separate files.
Move shared tables to a separate file as well.
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/Makefile | 40 | ||||
-rw-r--r-- | libavcodec/adpcm.c | 734 | ||||
-rw-r--r-- | libavcodec/adpcm.h | 46 | ||||
-rw-r--r-- | libavcodec/adpcm_data.c | 78 | ||||
-rw-r--r-- | libavcodec/adpcm_data.h | 37 | ||||
-rw-r--r-- | libavcodec/adpcmenc.c | 655 |
6 files changed, 861 insertions, 729 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 1bb6b09..7697f73 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -483,10 +483,10 @@ OBJS-$(CONFIG_PCM_U32LE_ENCODER) += pcm.o OBJS-$(CONFIG_PCM_ZORK_DECODER) += pcm.o OBJS-$(CONFIG_PCM_ZORK_ENCODER) += pcm.o -OBJS-$(CONFIG_ADPCM_4XM_DECODER) += adpcm.o +OBJS-$(CONFIG_ADPCM_4XM_DECODER) += adpcm.o adpcm_data.o OBJS-$(CONFIG_ADPCM_ADX_DECODER) += adxdec.o OBJS-$(CONFIG_ADPCM_ADX_ENCODER) += adxenc.o -OBJS-$(CONFIG_ADPCM_CT_DECODER) += adpcm.o +OBJS-$(CONFIG_ADPCM_CT_DECODER) += adpcm.o adpcm_data.o OBJS-$(CONFIG_ADPCM_EA_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_EA_R1_DECODER) += adpcm.o @@ -497,29 +497,29 @@ OBJS-$(CONFIG_ADPCM_G722_DECODER) += g722.o OBJS-$(CONFIG_ADPCM_G722_ENCODER) += g722.o OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o -OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_MS_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_MS_ENCODER) += adpcm.o +OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_EA_SEAD_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER) += adpcmenc.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER) += adpcmenc.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_MS_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_MS_ENCODER) += adpcmenc.o adpcm_data.o OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_SWF_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_SWF_ENCODER) += adpcm.o +OBJS-$(CONFIG_ADPCM_SWF_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_SWF_ENCODER) += adpcmenc.o adpcm_data.o OBJS-$(CONFIG_ADPCM_THP_DECODER) += adpcm.o OBJS-$(CONFIG_ADPCM_XA_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o -OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o +OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER) += adpcm.o adpcm_data.o +OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcmenc.o adpcm_data.o # libavformat dependencies OBJS-$(CONFIG_ADTS_MUXER) += mpeg4audio.o diff --git a/libavcodec/adpcm.c b/libavcodec/adpcm.c index 70a5360..c9ec0c3 100644 --- a/libavcodec/adpcm.c +++ b/libavcodec/adpcm.c @@ -1,5 +1,4 @@ /* - * ADPCM codecs * Copyright (c) 2001-2003 The ffmpeg Project * * This file is part of Libav. @@ -22,10 +21,12 @@ #include "get_bits.h" #include "put_bits.h" #include "bytestream.h" +#include "adpcm.h" +#include "adpcm_data.h" /** * @file - * ADPCM codecs. + * ADPCM decoders * First version by Francois Revol (revol@free.fr) * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood) * by Mike Melanson (melanson@pcisys.net) @@ -54,48 +55,6 @@ * 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 -}; - -/** Divided by 4 to fit in 8-bit integers */ -static const uint8_t AdaptCoeff1[] = { - 64, 128, 0, 48, 60, 115, 98 -}; - -/** Divided by 4 to fit in 8-bit integers */ -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 }, @@ -118,632 +77,15 @@ 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 TrellisPath { - int nibble; - int prev; -} TrellisPath; - -typedef struct TrellisNode { - uint32_t ssd; - int path; - int sample1; - int sample2; - int step; -} TrellisNode; - -typedef struct ADPCMContext { +typedef struct ADPCMDecodeContext { ADPCMChannelStatus status[6]; - TrellisPath *paths; - TrellisNode *node_buf; - TrellisNode **nodep_buf; - uint8_t *trellis_hash; -} ADPCMContext; - -#define FREEZE_INTERVAL 128 - -/* XXX: implement encoding */ - -#if CONFIG_ENCODERS -static av_cold int adpcm_encode_init(AVCodecContext *avctx) -{ - ADPCMContext *s = avctx->priv_data; - uint8_t *extradata; - int i; - 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; - } - - if (avctx->trellis) { - int frontier = 1 << avctx->trellis; - int max_paths = frontier * FREEZE_INTERVAL; - FF_ALLOC_OR_GOTO(avctx, s->paths, max_paths * sizeof(*s->paths), error); - FF_ALLOC_OR_GOTO(avctx, s->node_buf, 2 * frontier * sizeof(*s->node_buf), error); - FF_ALLOC_OR_GOTO(avctx, s->nodep_buf, 2 * frontier * sizeof(*s->nodep_buf), error); - FF_ALLOC_OR_GOTO(avctx, s->trellis_hash, 65536 * sizeof(*s->trellis_hash), error); - } - - 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; - avctx->extradata_size = 32; - extradata = avctx->extradata = av_malloc(avctx->extradata_size); - if (!extradata) - return AVERROR(ENOMEM); - bytestream_put_le16(&extradata, avctx->frame_size); - bytestream_put_le16(&extradata, 7); /* wNumCoef */ - for (i = 0; i < 7; i++) { - bytestream_put_le16(&extradata, AdaptCoeff1[i] * 4); - bytestream_put_le16(&extradata, AdaptCoeff2[i] * 4); - } - 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"); - goto error; - } - avctx->frame_size = 512 * (avctx->sample_rate / 11025); - break; - default: - goto error; - } - - avctx->coded_frame= avcodec_alloc_frame(); - avctx->coded_frame->key_frame= 1; - - return 0; -error: - av_freep(&s->paths); - av_freep(&s->node_buf); - av_freep(&s->nodep_buf); - av_freep(&s->trellis_hash); - return -1; -} - -static av_cold int adpcm_encode_close(AVCodecContext *avctx) -{ - ADPCMContext *s = avctx->priv_data; - av_freep(&avctx->coded_frame); - av_freep(&s->paths); - av_freep(&s->node_buf); - av_freep(&s->nodep_buf); - av_freep(&s->trellis_hash); - - 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; -} - -static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples, - uint8_t *dst, ADPCMChannelStatus *c, int n) -{ - //FIXME 6% faster if frontier is a compile-time constant - ADPCMContext *s = avctx->priv_data; - const int frontier = 1 << avctx->trellis; - const int stride = avctx->channels; - const int version = avctx->codec->id; - TrellisPath *paths = s->paths, *p; - TrellisNode *node_buf = s->node_buf; - TrellisNode **nodep_buf = s->nodep_buf; - TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd - TrellisNode **nodes_next = nodep_buf + frontier; - int pathn = 0, froze = -1, i, j, k, generation = 0; - uint8_t *hash = s->trellis_hash; - memset(hash, 0xff, 65536 * sizeof(*hash)); - - memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf)); - nodes[0] = node_buf + frontier; - 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; i<n; i++) { - TrellisNode *t = node_buf + frontier*(i&1); - TrellisNode **u; - int sample = samples[i*stride]; - int heap_pos = 0; - memset(nodes_next, 0, frontier*sizeof(TrellisNode*)); - for(j=0; j<frontier && nodes[j]; j++) { - // higher j have higher ssd already, so they're likely to yield 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)) / 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;\ - int pos;\ - TrellisNode *u;\ - uint8_t *h;\ - dec_sample = av_clip_int16(dec_sample);\ - d = sample - dec_sample;\ - ssd = nodes[j]->ssd + d*d;\ - /* Check for wraparound, skip such samples completely. \ - * Note, changing ssd to a 64 bit variable would be \ - * simpler, avoiding this check, but it's slower on \ - * x86 32 bit at the moment. */\ - if (ssd < nodes[j]->ssd)\ - goto next_##NAME;\ - /* 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. - * Since nodes in the previous generation are iterated - * through a heap, they're roughly ordered from better to - * worse, but not strictly ordered. Therefore, an earlier - * node with the same sample value is better in most cases - * (and thus the current is skipped), but not strictly - * in all cases. Only skipping samples where ssd >= - * ssd of the earlier node with the same sample gives - * slightly worse quality, though, for some reason. */ \ - h = &hash[(uint16_t) dec_sample];\ - if (*h == generation)\ - goto next_##NAME;\ - if (heap_pos < frontier) {\ - pos = heap_pos++;\ - } else {\ - /* Try to replace one of the leaf nodes with the new \ - * one, but try a different slot each time. */\ - pos = (frontier >> 1) + (heap_pos & ((frontier >> 1) - 1));\ - if (ssd > nodes_next[pos]->ssd)\ - goto next_##NAME;\ - heap_pos++;\ - }\ - *h = generation;\ - u = nodes_next[pos];\ - if(!u) {\ - assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\ - u = t++;\ - nodes_next[pos] = u;\ - 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;\ - /* Sift the newly inserted node up in the heap to \ - * restore the heap property. */\ - while (pos > 0) {\ - int parent = (pos - 1) >> 1;\ - if (nodes_next[parent]->ssd <= ssd)\ - break;\ - FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\ - pos = parent;\ - }\ - 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; - - generation++; - if (generation == 255) { - memset(hash, 0xff, 65536 * sizeof(*hash)); - generation = 0; - } - - // prevent overflow - if(nodes[0]->ssd > (1<<28)) { - for(j=1; j<frontier && nodes[j]; j++) - nodes[j]->ssd -= nodes[0]->ssd; - nodes[0]->ssd = 0; - } - - // merge old paths to save memory - if(i == froze + FREEZE_INTERVAL) { - p = &paths[nodes[0]->path]; - for(k=i; k>froze; k--) { - dst[k] = p->nibble; - p = &paths[p->prev]; - } - froze = i; - pathn = 0; - // other nodes might use paths that don't coincide with the frozen one. - // checking which nodes do so is too slow, so just kill them all. - // this also slightly improves quality, but I don't know why. - memset(nodes+1, 0, (frontier-1)*sizeof(TrellisNode*)); - } - } - - p = &paths[nodes[0]->path]; - for(i=n-1; i>froze; i--) { - dst[i] = p->nibble; - p = &paths[p->prev]; - } - - c->predictor = nodes[0]->sample1; - c->sample1 = nodes[0]->sample1; - c->sample2 = nodes[0]->sample2; - c->step_index = nodes[0]->step; - c->step = nodes[0]->step; - c->idelta = nodes[0]->step; -} - -static int adpcm_encode_frame(AVCodecContext *avctx, - unsigned char *frame, int buf_size, void *data) -{ - int n, i, st; - short *samples; - unsigned char *dst; - ADPCMContext *c = avctx->priv_data; - uint8_t *buf; - - dst = frame; - samples = (short *)data; - st= avctx->channels == 2; -/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */ - - switch(avctx->codec->id) { - case CODEC_ID_ADPCM_IMA_WAV: - n = avctx->frame_size / 8; - c->status[0].prev_sample = (signed short)samples[0]; /* XXX */ -/* c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */ - bytestream_put_le16(&dst, c->status[0].prev_sample); - *dst++ = (unsigned char)c->status[0].step_index; - *dst++ = 0; /* unknown */ - samples++; - if (avctx->channels == 2) { - c->status[1].prev_sample = (signed short)samples[0]; -/* c->status[1].step_index = 0; */ - bytestream_put_le16(&dst, c->status[1].prev_sample); - *dst++ = (unsigned char)c->status[1].step_index; - *dst++ = 0; - samples++; - } - - /* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */ - if(avctx->trellis > 0) { - FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error); - adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8); - if(avctx->channels == 2) - adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8); - for(i=0; i<n; i++) { - *dst++ = buf[8*i+0] | (buf[8*i+1] << 4); - *dst++ = buf[8*i+2] | (buf[8*i+3] << 4); - *dst++ = buf[8*i+4] | (buf[8*i+5] << 4); - *dst++ = buf[8*i+6] | (buf[8*i+7] << 4); - if (avctx->channels == 2) { - uint8_t *buf1 = buf + n*8; - *dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4); - *dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4); - *dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4); - *dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4); - } - } - av_free(buf); - } else - for (; n>0; n--) { - *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]); - *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4; - dst++; - *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]); - *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4; - dst++; - *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]); - *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4; - dst++; - *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]); - *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4; - dst++; - /* right channel */ - if (avctx->channels == 2) { - *dst = adpcm_ima_compress_sample(&c->status[1], samples[1]); - *dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4; - dst++; - *dst = adpcm_ima_compress_sample(&c->status[1], samples[5]); - *dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4; - dst++; - *dst = adpcm_ima_compress_sample(&c->status[1], samples[9]); - *dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4; - dst++; - *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]); - *dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4; - dst++; - } - samples += 8 * avctx->channels; - } - break; - case CODEC_ID_ADPCM_IMA_QT: - { - int ch, i; - PutBitContext pb; - init_put_bits(&pb, dst, buf_size*8); - - for(ch=0; ch<avctx->channels; ch++){ - put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7); - put_bits(&pb, 7, c->status[ch].step_index); - if(avctx->trellis > 0) { - uint8_t buf[64]; - adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64); - for(i=0; i<64; i++) - put_bits(&pb, 4, buf[i^1]); - c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F; - } else { - for (i=0; i<64; i+=2){ - int t1, t2; - t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]); - t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]); - put_bits(&pb, 4, t2); - put_bits(&pb, 4, t1); - } - c->status[ch].prev_sample &= ~0x7F; - } - } - - flush_put_bits(&pb); - dst += put_bits_count(&pb)>>3; - break; - } - case CODEC_ID_ADPCM_SWF: - { - int i; - PutBitContext pb; - init_put_bits(&pb, dst, buf_size*8); - - n = avctx->frame_size-1; - - //Store AdpcmCodeSize - put_bits(&pb, 2, 2); //Set 4bits flash adpcm format - - //Init the encoder state - for(i=0; i<avctx->channels; i++){ - c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits - put_sbits(&pb, 16, samples[i]); - put_bits(&pb, 6, c->status[i].step_index); - c->status[i].prev_sample = (signed short)samples[i]; - } - - if(avctx->trellis > 0) { - FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error); - adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n); - if (avctx->channels == 2) - adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n); - for(i=0; i<n; i++) { - put_bits(&pb, 4, buf[i]); - if (avctx->channels == 2) - put_bits(&pb, 4, buf[n+i]); - } - av_free(buf); - } else { - for (i=1; i<avctx->frame_size; i++) { - put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i])); - if (avctx->channels == 2) - put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1])); - } - } - flush_put_bits(&pb); - dst += put_bits_count(&pb)>>3; - break; - } - case CODEC_ID_ADPCM_MS: - for(i=0; i<avctx->channels; i++){ - int predictor=0; - - *dst++ = predictor; - c->status[i].coeff1 = AdaptCoeff1[predictor]; - c->status[i].coeff2 = AdaptCoeff2[predictor]; - } - for(i=0; i<avctx->channels; i++){ - if (c->status[i].idelta < 16) - c->status[i].idelta = 16; - - bytestream_put_le16(&dst, c->status[i].idelta); - } - for(i=0; i<avctx->channels; i++){ - c->status[i].sample2= *samples++; - } - for(i=0; i<avctx->channels; i++){ - c->status[i].sample1= *samples++; - - bytestream_put_le16(&dst, c->status[i].sample1); - } - for(i=0; i<avctx->channels; i++) - bytestream_put_le16(&dst, c->status[i].sample2); - - if(avctx->trellis > 0) { - int n = avctx->block_align - 7*avctx->channels; - FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error); - if(avctx->channels == 1) { - adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); - for(i=0; i<n; i+=2) - *dst++ = (buf[i] << 4) | buf[i+1]; - } else { - adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); - adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n); - for(i=0; i<n; i++) - *dst++ = (buf[i] << 4) | buf[n+i]; - } - av_free(buf); - } else - for(i=7*avctx->channels; i<avctx->block_align; i++) { - int nibble; - nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++)<<4; - nibble|= adpcm_ms_compress_sample(&c->status[st], *samples++); - *dst++ = nibble; - } - break; - case CODEC_ID_ADPCM_YAMAHA: - n = avctx->frame_size / 2; - if(avctx->trellis > 0) { - FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error); - n *= 2; - if(avctx->channels == 1) { - adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); - for(i=0; i<n; i+=2) - *dst++ = buf[i] | (buf[i+1] << 4); - } else { - adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); - adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n); - for(i=0; i<n; i++) - *dst++ = buf[i] | (buf[n+i] << 4); - } - av_free(buf); - } else - for (n *= avctx->channels; n>0; n--) { - int nibble; - nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++); - nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4; - *dst++ = nibble; - } - break; - default: - error: - return -1; - } - return dst - frame; -} -#endif //CONFIG_ENCODERS +} ADPCMDecodeContext; static av_cold int adpcm_decode_init(AVCodecContext * avctx) { - ADPCMContext *c = avctx->priv_data; + ADPCMDecodeContext *c = avctx->priv_data; unsigned int max_channels = 2; switch(avctx->codec->id) { @@ -786,8 +128,8 @@ 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; @@ -816,7 +158,7 @@ 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; @@ -837,7 +179,7 @@ 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; @@ -870,9 +212,9 @@ 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; } @@ -964,7 +306,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, { 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]; @@ -1030,7 +372,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, cs->step_index = 88; } - cs->step = step_table[cs->step_index]; + cs->step = ff_adpcm_step_table[cs->step_index]; samples = (short*)data + channel; @@ -1114,10 +456,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx, 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].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[0]]; + c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[0]]; + c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor[1]]; + c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor[1]]; c->status[0].sample1 = bytestream_get_le16(&src); if (st) c->status[1].sample1 = bytestream_get_le16(&src); @@ -1586,7 +928,7 @@ 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; @@ -1705,44 +1047,18 @@ static int adpcm_decode_frame(AVCodecContext *avctx, } - -#if CONFIG_ENCODERS -#define ADPCM_ENCODER(id,name,long_name_) \ -AVCodec ff_ ## name ## _encoder = { \ - #name, \ - AVMEDIA_TYPE_AUDIO, \ - id, \ - sizeof(ADPCMContext), \ - adpcm_encode_init, \ - adpcm_encode_frame, \ - adpcm_encode_close, \ - NULL, \ - .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \ - .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ -} -#else -#define ADPCM_ENCODER(id,name,long_name_) -#endif - -#if CONFIG_DECODERS #define ADPCM_DECODER(id,name,long_name_) \ AVCodec ff_ ## name ## _decoder = { \ #name, \ AVMEDIA_TYPE_AUDIO, \ id, \ - sizeof(ADPCMContext), \ + sizeof(ADPCMDecodeContext), \ adpcm_decode_init, \ NULL, \ NULL, \ adpcm_decode_frame, \ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ } -#else -#define ADPCM_DECODER(id,name,long_name_) -#endif - -#define ADPCM_CODEC(id,name,long_name_) \ - ADPCM_ENCODER(id,name,long_name_); ADPCM_DECODER(id,name,long_name_) /* Note: Do not forget to add new entries to the Makefile as well. */ ADPCM_DECODER(CODEC_ID_ADPCM_4XM, adpcm_4xm, "ADPCM 4X Movie"); @@ -1759,15 +1075,15 @@ 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"); diff --git a/libavcodec/adpcm.h b/libavcodec/adpcm.h new file mode 100644 index 0000000..aed5048 --- /dev/null +++ b/libavcodec/adpcm.h @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2001-2003 The ffmpeg Project + * + * This file is part of Libav. + * + * 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. + * + * 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * ADPCM encoder/decoder common header. + */ + +#ifndef AVCODEC_ADPCM_H +#define AVCODEC_ADPCM_H + +#define BLKSIZE 1024 + +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; + +#endif /* AVCODEC_ADPCM_H */ diff --git a/libavcodec/adpcm_data.c b/libavcodec/adpcm_data.c new file mode 100644 index 0000000..9dc5670 --- /dev/null +++ b/libavcodec/adpcm_data.c @@ -0,0 +1,78 @@ +/* + * Copyright (c) 2001-2003 The ffmpeg Project + * + * This file is part of Libav. + * + * 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. + * + * 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * ADPCM tables + */ + +#include <stdint.h> + +/* ff_adpcm_step_table[] and ff_adpcm_index_table[] are from the ADPCM + reference source */ +/* This is the index table: */ +const int8_t ff_adpcm_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: + */ +const int16_t ff_adpcm_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 */ +/* ff_adpcm_AdaptationTable[], ff_adpcm_AdaptCoeff1[], and + ff_adpcm_AdaptCoeff2[] are from libsndfile */ +const int16_t ff_adpcm_AdaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 +}; + +/** Divided by 4 to fit in 8-bit integers */ +const uint8_t ff_adpcm_AdaptCoeff1[] = { + 64, 128, 0, 48, 60, 115, 98 +}; + +/** Divided by 4 to fit in 8-bit integers */ +const int8_t ff_adpcm_AdaptCoeff2[] = { + 0, -64, 0, 16, 0, -52, -58 +}; + +const int16_t ff_adpcm_yamaha_indexscale[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 230, 230, 230, 230, 307, 409, 512, 614 +}; + +const int8_t ff_adpcm_yamaha_difflookup[] = { + 1, 3, 5, 7, 9, 11, 13, 15, + -1, -3, -5, -7, -9, -11, -13, -15 +}; diff --git a/libavcodec/adpcm_data.h b/libavcodec/adpcm_data.h new file mode 100644 index 0000000..baca426 --- /dev/null +++ b/libavcodec/adpcm_data.h @@ -0,0 +1,37 @@ +/* + * Copyright (c) 2001-2003 The ffmpeg Project + * + * This file is part of Libav. + * + * 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. + * + * 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * ADPCM tables + */ + +#ifndef AVCODEC_ADPCM_DATA_H +#define AVCODEC_ADPCM_DATA_H + +extern const int8_t ff_adpcm_index_table[16]; +extern const int16_t ff_adpcm_step_table[89]; +extern const int16_t ff_adpcm_AdaptationTable[]; +extern const uint8_t ff_adpcm_AdaptCoeff1[]; +extern const int8_t ff_adpcm_AdaptCoeff2[]; +extern const int16_t ff_adpcm_yamaha_indexscale[]; +extern const int8_t ff_adpcm_yamaha_difflookup[]; + +#endif /* AVCODEC_ADPCM_DATA_H */ diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c new file mode 100644 index 0000000..ec06284 --- /dev/null +++ b/libavcodec/adpcmenc.c @@ -0,0 +1,655 @@ +/* + * Copyright (c) 2001-2003 The ffmpeg Project + * + * This file is part of Libav. + * + * 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. + * + * 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "avcodec.h" +#include "get_bits.h" +#include "put_bits.h" +#include "bytestream.h" +#include "adpcm.h" +#include "adpcm_data.h" + +/** + * @file + * ADPCM encoders + * First version by Francois Revol (revol@free.fr) + * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood) + * by Mike Melanson (melanson@pcisys.net) + * + * 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 + */ + +typedef struct TrellisPath { + int nibble; + int prev; +} TrellisPath; + +typedef struct TrellisNode { + uint32_t ssd; + int path; + int sample1; + int sample2; + int step; +} TrellisNode; + +typedef struct ADPCMEncodeContext { + ADPCMChannelStatus status[6]; + TrellisPath *paths; + TrellisNode *node_buf; + TrellisNode **nodep_buf; + uint8_t *trellis_hash; +} ADPCMEncodeContext; + +#define FREEZE_INTERVAL 128 + +static av_cold int adpcm_encode_init(AVCodecContext *avctx) +{ + ADPCMEncodeContext *s = avctx->priv_data; + uint8_t *extradata; + int i; + 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; + } + + if (avctx->trellis) { + int frontier = 1 << avctx->trellis; + int max_paths = frontier * FREEZE_INTERVAL; + FF_ALLOC_OR_GOTO(avctx, s->paths, max_paths * sizeof(*s->paths), error); + FF_ALLOC_OR_GOTO(avctx, s->node_buf, 2 * frontier * sizeof(*s->node_buf), error); + FF_ALLOC_OR_GOTO(avctx, s->nodep_buf, 2 * frontier * sizeof(*s->nodep_buf), error); + FF_ALLOC_OR_GOTO(avctx, s->trellis_hash, 65536 * sizeof(*s->trellis_hash), error); + } + + 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; + avctx->extradata_size = 32; + extradata = avctx->extradata = av_malloc(avctx->extradata_size); + if (!extradata) + return AVERROR(ENOMEM); + bytestream_put_le16(&extradata, avctx->frame_size); + bytestream_put_le16(&extradata, 7); /* wNumCoef */ + for (i = 0; i < 7; i++) { + bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4); + bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4); + } + 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"); + goto error; + } + avctx->frame_size = 512 * (avctx->sample_rate / 11025); + break; + default: + goto error; + } + + avctx->coded_frame= avcodec_alloc_frame(); + avctx->coded_frame->key_frame= 1; + + return 0; +error: + av_freep(&s->paths); + av_freep(&s->node_buf); + av_freep(&s->nodep_buf); + av_freep(&s->trellis_hash); + return -1; +} + +static av_cold int adpcm_encode_close(AVCodecContext *avctx) +{ + ADPCMEncodeContext *s = avctx->priv_data; + av_freep(&avctx->coded_frame); + av_freep(&s->paths); + av_freep(&s->node_buf); + av_freep(&s->nodep_buf); + av_freep(&s->trellis_hash); + + 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/ff_adpcm_step_table[c->step_index]) + (delta<0)*8; + c->prev_sample += ((ff_adpcm_step_table[c->step_index] * ff_adpcm_yamaha_difflookup[nibble]) / 8); + c->prev_sample = av_clip_int16(c->prev_sample); + c->step_index = av_clip(c->step_index + ff_adpcm_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 = (ff_adpcm_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 * ff_adpcm_yamaha_difflookup[nibble]) / 8); + c->predictor = av_clip_int16(c->predictor); + c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8; + c->step = av_clip(c->step, 127, 24567); + + return nibble; +} + +static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples, + uint8_t *dst, ADPCMChannelStatus *c, int n) +{ + //FIXME 6% faster if frontier is a compile-time constant + ADPCMEncodeContext *s = avctx->priv_data; + const int frontier = 1 << avctx->trellis; + const int stride = avctx->channels; + const int version = avctx->codec->id; + TrellisPath *paths = s->paths, *p; + TrellisNode *node_buf = s->node_buf; + TrellisNode **nodep_buf = s->nodep_buf; + TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd + TrellisNode **nodes_next = nodep_buf + frontier; + int pathn = 0, froze = -1, i, j, k, generation = 0; + uint8_t *hash = s->trellis_hash; + memset(hash, 0xff, 65536 * sizeof(*hash)); + + memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf)); + nodes[0] = node_buf + frontier; + 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; i<n; i++) { + TrellisNode *t = node_buf + frontier*(i&1); + TrellisNode **u; + int sample = samples[i*stride]; + int heap_pos = 0; + memset(nodes_next, 0, frontier*sizeof(TrellisNode*)); + for(j=0; j<frontier && nodes[j]; j++) { + // higher j have higher ssd already, so they're likely to yield 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)) / 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;\ + int pos;\ + TrellisNode *u;\ + uint8_t *h;\ + dec_sample = av_clip_int16(dec_sample);\ + d = sample - dec_sample;\ + ssd = nodes[j]->ssd + d*d;\ + /* Check for wraparound, skip such samples completely. \ + * Note, changing ssd to a 64 bit variable would be \ + * simpler, avoiding this check, but it's slower on \ + * x86 32 bit at the moment. */\ + if (ssd < nodes[j]->ssd)\ + goto next_##NAME;\ + /* 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. + * Since nodes in the previous generation are iterated + * through a heap, they're roughly ordered from better to + * worse, but not strictly ordered. Therefore, an earlier + * node with the same sample value is better in most cases + * (and thus the current is skipped), but not strictly + * in all cases. Only skipping samples where ssd >= + * ssd of the earlier node with the same sample gives + * slightly worse quality, though, for some reason. */ \ + h = &hash[(uint16_t) dec_sample];\ + if (*h == generation)\ + goto next_##NAME;\ + if (heap_pos < frontier) {\ + pos = heap_pos++;\ + } else {\ + /* Try to replace one of the leaf nodes with the new \ + * one, but try a different slot each time. */\ + pos = (frontier >> 1) + (heap_pos & ((frontier >> 1) - 1));\ + if (ssd > nodes_next[pos]->ssd)\ + goto next_##NAME;\ + heap_pos++;\ + }\ + *h = generation;\ + u = nodes_next[pos];\ + if(!u) {\ + assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\ + u = t++;\ + nodes_next[pos] = u;\ + 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;\ + /* Sift the newly inserted node up in the heap to \ + * restore the heap property. */\ + while (pos > 0) {\ + int parent = (pos - 1) >> 1;\ + if (nodes_next[parent]->ssd <= ssd)\ + break;\ + FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\ + pos = parent;\ + }\ + next_##NAME:; + STORE_NODE(ms, FFMAX(16, (ff_adpcm_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 * ff_adpcm_yamaha_difflookup[nibble]) / 8;\ + STORE_NODE(NAME, STEP_INDEX);\ + } + LOOP_NODES(ima, ff_adpcm_step_table[step], av_clip(step + ff_adpcm_index_table[nibble], 0, 88)); + } else { //CODEC_ID_ADPCM_YAMAHA + LOOP_NODES(yamaha, step, av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8, 127, 24567)); +#undef LOOP_NODES +#undef STORE_NODE + } + } + + u = nodes; + nodes = nodes_next; + nodes_next = u; + + generation++; + if (generation == 255) { + memset(hash, 0xff, 65536 * sizeof(*hash)); + generation = 0; + } + + // prevent overflow + if(nodes[0]->ssd > (1<<28)) { + for(j=1; j<frontier && nodes[j]; j++) + nodes[j]->ssd -= nodes[0]->ssd; + nodes[0]->ssd = 0; + } + + // merge old paths to save memory + if(i == froze + FREEZE_INTERVAL) { + p = &paths[nodes[0]->path]; + for(k=i; k>froze; k--) { + dst[k] = p->nibble; + p = &paths[p->prev]; + } + froze = i; + pathn = 0; + // other nodes might use paths that don't coincide with the frozen one. + // checking which nodes do so is too slow, so just kill them all. + // this also slightly improves quality, but I don't know why. + memset(nodes+1, 0, (frontier-1)*sizeof(TrellisNode*)); + } + } + + p = &paths[nodes[0]->path]; + for(i=n-1; i>froze; i--) { + dst[i] = p->nibble; + p = &paths[p->prev]; + } + + c->predictor = nodes[0]->sample1; + c->sample1 = nodes[0]->sample1; + c->sample2 = nodes[0]->sample2; + c->step_index = nodes[0]->step; + c->step = nodes[0]->step; + c->idelta = nodes[0]->step; +} + +static int adpcm_encode_frame(AVCodecContext *avctx, + unsigned char *frame, int buf_size, void *data) +{ + int n, i, st; + short *samples; + unsigned char *dst; + ADPCMEncodeContext *c = avctx->priv_data; + uint8_t *buf; + + dst = frame; + samples = (short *)data; + st= avctx->channels == 2; +/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */ + + switch(avctx->codec->id) { + case CODEC_ID_ADPCM_IMA_WAV: + n = avctx->frame_size / 8; + c->status[0].prev_sample = (signed short)samples[0]; /* XXX */ +/* c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */ + bytestream_put_le16(&dst, c->status[0].prev_sample); + *dst++ = (unsigned char)c->status[0].step_index; + *dst++ = 0; /* unknown */ + samples++; + if (avctx->channels == 2) { + c->status[1].prev_sample = (signed short)samples[0]; +/* c->status[1].step_index = 0; */ + bytestream_put_le16(&dst, c->status[1].prev_sample); + *dst++ = (unsigned char)c->status[1].step_index; + *dst++ = 0; + samples++; + } + + /* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */ + if(avctx->trellis > 0) { + FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error); + adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8); + if(avctx->channels == 2) + adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8); + for(i=0; i<n; i++) { + *dst++ = buf[8*i+0] | (buf[8*i+1] << 4); + *dst++ = buf[8*i+2] | (buf[8*i+3] << 4); + *dst++ = buf[8*i+4] | (buf[8*i+5] << 4); + *dst++ = buf[8*i+6] | (buf[8*i+7] << 4); + if (avctx->channels == 2) { + uint8_t *buf1 = buf + n*8; + *dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4); + *dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4); + *dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4); + *dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4); + } + } + av_free(buf); + } else + for (; n>0; n--) { + *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]); + *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4; + dst++; + *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]); + *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4; + dst++; + *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]); + *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4; + dst++; + *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]); + *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4; + dst++; + /* right channel */ + if (avctx->channels == 2) { + *dst = adpcm_ima_compress_sample(&c->status[1], samples[1]); + *dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4; + dst++; + *dst = adpcm_ima_compress_sample(&c->status[1], samples[5]); + *dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4; + dst++; + *dst = adpcm_ima_compress_sample(&c->status[1], samples[9]); + *dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4; + dst++; + *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]); + *dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4; + dst++; + } + samples += 8 * avctx->channels; + } + break; + case CODEC_ID_ADPCM_IMA_QT: + { + int ch, i; + PutBitContext pb; + init_put_bits(&pb, dst, buf_size*8); + + for(ch=0; ch<avctx->channels; ch++){ + put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7); + put_bits(&pb, 7, c->status[ch].step_index); + if(avctx->trellis > 0) { + uint8_t buf[64]; + adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64); + for(i=0; i<64; i++) + put_bits(&pb, 4, buf[i^1]); + c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F; + } else { + for (i=0; i<64; i+=2){ + int t1, t2; + t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]); + t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]); + put_bits(&pb, 4, t2); + put_bits(&pb, 4, t1); + } + c->status[ch].prev_sample &= ~0x7F; + } + } + + flush_put_bits(&pb); + dst += put_bits_count(&pb)>>3; + break; + } + case CODEC_ID_ADPCM_SWF: + { + int i; + PutBitContext pb; + init_put_bits(&pb, dst, buf_size*8); + + n = avctx->frame_size-1; + + //Store AdpcmCodeSize + put_bits(&pb, 2, 2); //Set 4bits flash adpcm format + + //Init the encoder state + for(i=0; i<avctx->channels; i++){ + c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits + put_sbits(&pb, 16, samples[i]); + put_bits(&pb, 6, c->status[i].step_index); + c->status[i].prev_sample = (signed short)samples[i]; + } + + if(avctx->trellis > 0) { + FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error); + adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n); + if (avctx->channels == 2) + adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n); + for(i=0; i<n; i++) { + put_bits(&pb, 4, buf[i]); + if (avctx->channels == 2) + put_bits(&pb, 4, buf[n+i]); + } + av_free(buf); + } else { + for (i=1; i<avctx->frame_size; i++) { + put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i])); + if (avctx->channels == 2) + put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1])); + } + } + flush_put_bits(&pb); + dst += put_bits_count(&pb)>>3; + break; + } + case CODEC_ID_ADPCM_MS: + for(i=0; i<avctx->channels; i++){ + int predictor=0; + + *dst++ = predictor; + c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor]; + c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor]; + } + for(i=0; i<avctx->channels; i++){ + if (c->status[i].idelta < 16) + c->status[i].idelta = 16; + + bytestream_put_le16(&dst, c->status[i].idelta); + } + for(i=0; i<avctx->channels; i++){ + c->status[i].sample2= *samples++; + } + for(i=0; i<avctx->channels; i++){ + c->status[i].sample1= *samples++; + + bytestream_put_le16(&dst, c->status[i].sample1); + } + for(i=0; i<avctx->channels; i++) + bytestream_put_le16(&dst, c->status[i].sample2); + + if(avctx->trellis > 0) { + int n = avctx->block_align - 7*avctx->channels; + FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error); + if(avctx->channels == 1) { + adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); + for(i=0; i<n; i+=2) + *dst++ = (buf[i] << 4) | buf[i+1]; + } else { + adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); + adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n); + for(i=0; i<n; i++) + *dst++ = (buf[i] << 4) | buf[n+i]; + } + av_free(buf); + } else + for(i=7*avctx->channels; i<avctx->block_align; i++) { + int nibble; + nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++)<<4; + nibble|= adpcm_ms_compress_sample(&c->status[st], *samples++); + *dst++ = nibble; + } + break; + case CODEC_ID_ADPCM_YAMAHA: + n = avctx->frame_size / 2; + if(avctx->trellis > 0) { + FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error); + n *= 2; + if(avctx->channels == 1) { + adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); + for(i=0; i<n; i+=2) + *dst++ = buf[i] | (buf[i+1] << 4); + } else { + adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n); + adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n); + for(i=0; i<n; i++) + *dst++ = buf[i] | (buf[n+i] << 4); + } + av_free(buf); + } else + for (n *= avctx->channels; n>0; n--) { + int nibble; + nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++); + nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4; + *dst++ = nibble; + } + break; + default: + error: + return -1; + } + return dst - frame; +} + + +#define ADPCM_ENCODER(id,name,long_name_) \ +AVCodec ff_ ## name ## _encoder = { \ + #name, \ + AVMEDIA_TYPE_AUDIO, \ + id, \ + sizeof(ADPCMEncodeContext), \ + adpcm_encode_init, \ + adpcm_encode_frame, \ + adpcm_encode_close, \ + NULL, \ + .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \ + .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ +} + +ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime"); +ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV"); +ADPCM_ENCODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft"); +ADPCM_ENCODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash"); +ADPCM_ENCODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha"); |