[FFmpeg-cvslog] adpcm: split ADPCM encoders and decoders into separate files.
Justin Ruggles
git at videolan.org
Mon Sep 12 21:02:33 CEST 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Wed Sep 7 18:34:09 2011 -0400| [826c56d16e55f3819a75d01f957dd295aa1e9f3a] | committer: Justin Ruggles
adpcm: split ADPCM encoders and decoders into separate files.
Move shared tables to a separate file as well.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=826c56d16e55f3819a75d01f957dd295aa1e9f3a
---
libavcodec/Makefile | 40 ++--
libavcodec/adpcm.c | 734 ++---------------------------------------------
libavcodec/adpcm.h | 46 +++
libavcodec/adpcm_data.c | 78 +++++
libavcodec/adpcm_data.h | 37 +++
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 at free.fr)
* Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
* by Mike Melanson (melanson at 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 at free.fr)
+ * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
+ * by Mike Melanson (melanson at 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");
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