Merge remote-tracking branch 'qatar/master'

* qatar/master:
  adpcm: split ADPCM encoders and decoders into separate files.
  doc/avconv: fix typo.
  rv34: check that subsequent slices have the same type as first one.
  smacker demuxer: handle possible av_realloc() failure.
  lavfi: add split filter from soc.
  lavfi: add showinfo filter
  libxavs: add private options corresponding to deprecated global options

Conflicts:
	Changelog
	libavcodec/adpcm.c
	libavfilter/avfilter.h
	libavfilter/vf_showinfo.c
	libavfilter/vf_split.c
	libavformat/smacker.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

8 files changed, 1004 insertions, 793 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 741a9b3..17cc1b4 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -499,10 +499,10 @@ 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-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o
@@ -513,29 +513,29 @@ OBJS-$(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 4e12993..77e1ef6 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 FFmpeg.
@@ -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 },
@@ -118,668 +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;
-        avctx->bits_per_coded_sample = 4;
-        /* 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->bits_per_coded_sample = 4;
-        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_ima_qt_compress_sample(ADPCMChannelStatus *c, short sample)
-{
-    int delta = sample - c->prev_sample;
-    int diff, step = step_table[c->step_index];
-    int nibble = 8*(delta < 0);
-
-    delta= abs(delta);
-    diff = delta + (step >> 3);
-
-    if (delta >= step) {
-        nibble |= 4;
-        delta -= step;
-    }
-    step >>= 1;
-    if (delta >= step) {
-        nibble |= 2;
-        delta -= step;
-    }
-    step >>= 1;
-    if (delta >= step) {
-        nibble |= 1;
-        delta -= step;
-    }
-    diff -= delta;
-
-    if (nibble & 8)
-        c->prev_sample -= diff;
-    else
-        c->prev_sample += diff;
-
-    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]);
-            } else {
-                for (i=0; i<64; i+=2){
-                    int t1, t2;
-                    t1 = adpcm_ima_qt_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
-                    t2 = adpcm_ima_qt_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
-                    put_bits(&pb, 4, t2);
-                    put_bits(&pb, 4, t1);
-                }
-            }
-        }
-
-        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) {
@@ -823,8 +129,8 @@ 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;
 
@@ -850,8 +156,8 @@ static inline int adpcm_ima_qt_expand_nibble(ADPCMChannelStatus *c, int nibble,
     int predictor;
     int diff, step;
 
-    step = step_table[c->step_index];
-    step_index = c->step_index + index_table[nibble];
+    step = ff_adpcm_step_table[c->step_index];
+    step_index = c->step_index + ff_adpcm_index_table[nibble];
     step_index = av_clip(step_index, 0, 88);
 
     diff = step >> 3;
@@ -879,7 +185,7 @@ 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;
@@ -900,7 +206,7 @@ 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;
@@ -933,9 +239,9 @@ 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;
 }
@@ -1027,7 +333,7 @@ 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];
@@ -1183,10 +489,10 @@ 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);
@@ -1655,7 +961,7 @@ 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;
 
@@ -1774,44 +1080,18 @@ 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");
@@ -1828,15 +1108,15 @@ 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..475719a
--- /dev/null
+++ b/libavcodec/adpcm.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2001-2003 The ffmpeg Project
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @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..3654a8d
--- /dev/null
+++ b/libavcodec/adpcm_data.c
@@ -0,0 +1,78 @@
+/*
+ * Copyright (c) 2001-2003 The ffmpeg Project
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @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..2aa7df5
--- /dev/null
+++ b/libavcodec/adpcm_data.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright (c) 2001-2003 The ffmpeg Project
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @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..c39bab8
--- /dev/null
+++ b/libavcodec/adpcmenc.c
@@ -0,0 +1,691 @@
+/*
+ * Copyright (c) 2001-2003 The ffmpeg Project
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avcodec.h"
+#include "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;
+        avctx->bits_per_coded_sample = 4;
+        /* 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->bits_per_coded_sample = 4;
+        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_ima_qt_compress_sample(ADPCMChannelStatus *c, short sample)
+{
+    int delta = sample - c->prev_sample;
+    int diff, step = ff_adpcm_step_table[c->step_index];
+    int nibble = 8*(delta < 0);
+
+    delta= abs(delta);
+    diff = delta + (step >> 3);
+
+    if (delta >= step) {
+        nibble |= 4;
+        delta -= step;
+    }
+    step >>= 1;
+    if (delta >= step) {
+        nibble |= 2;
+        delta -= step;
+    }
+    step >>= 1;
+    if (delta >= step) {
+        nibble |= 1;
+        delta -= step;
+    }
+    diff -= delta;
+
+    if (nibble & 8)
+        c->prev_sample -= diff;
+    else
+        c->prev_sample += diff;
+
+    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]);
+            } else {
+                for (i=0; i<64; i+=2){
+                    int t1, t2;
+                    t1 = adpcm_ima_qt_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
+                    t2 = adpcm_ima_qt_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
+                    put_bits(&pb, 4, t2);
+                    put_bits(&pb, 4, t1);
+                }
+            }
+        }
+
+        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");
diff --git a/libavcodec/libxavs.c b/libavcodec/libxavs.c
index 4b604bd..3a35a46 100644
--- a/libavcodec/libxavs.c
+++ b/libavcodec/libxavs.c
@@ -24,8 +24,11 @@
 #include <string.h>
 #include <math.h>
 #include <stdint.h>
+#include <float.h>
 #include <xavs.h>
 #include "avcodec.h"
+#include "internal.h"
+#include "libavutil/opt.h"
 
 #define END_OF_STREAM 0x001
 
@@ -41,6 +44,15 @@ typedef struct XavsContext {
     int             sei_size;
     AVFrame         out_pic;
     int             end_of_stream;
+    float crf;
+    int cqp;
+    int b_bias;
+    float cplxblur;
+    int direct_pred;
+    int aud;
+    int fast_pskip;
+    int mbtree;
+    int mixed_refs;
 } XavsContext;
 
 static void XAVS_log(void *p, int level, const char *fmt, va_list args)
@@ -181,13 +193,17 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
     x4->params.pf_log               = XAVS_log;
     x4->params.p_log_private        = avctx;
     x4->params.i_keyint_max         = avctx->gop_size;
-    x4->params.rc.i_bitrate         = avctx->bit_rate       / 1000;
+    if (avctx->bit_rate) {
+        x4->params.rc.i_bitrate   = avctx->bit_rate / 1000;
+        x4->params.rc.i_rc_method = XAVS_RC_ABR;
+    }
     x4->params.rc.i_vbv_buffer_size = avctx->rc_buffer_size / 1000;
     x4->params.rc.i_vbv_max_bitrate = avctx->rc_max_rate    / 1000;
     x4->params.rc.b_stat_write      = avctx->flags & CODEC_FLAG_PASS1;
     if (avctx->flags & CODEC_FLAG_PASS2) {
         x4->params.rc.b_stat_read = 1;
     } else {
+#if FF_API_X264_GLOBAL_OPTS
         if (avctx->crf) {
             x4->params.rc.i_rc_method   = XAVS_RC_CRF;
             x4->params.rc.f_rf_constant = avctx->crf;
@@ -195,19 +211,63 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
             x4->params.rc.i_rc_method   = XAVS_RC_CQP;
             x4->params.rc.i_qp_constant = avctx->cqp;
         }
+#endif
+
+        if (x4->crf >= 0) {
+            x4->params.rc.i_rc_method   = XAVS_RC_CRF;
+            x4->params.rc.f_rf_constant = x4->crf;
+        } else if (x4->cqp >= 0) {
+            x4->params.rc.i_rc_method   = XAVS_RC_CQP;
+            x4->params.rc.i_qp_constant = x4->cqp;
+        }
     }
 
-    /* if neither crf nor cqp modes are selected we have to enable the RC */
-    /* we do it this way because we cannot check if the bitrate has been set */
-    if (!(avctx->crf || (avctx->cqp > -1)))
-        x4->params.rc.i_rc_method = XAVS_RC_ABR;
+#if FF_API_X264_GLOBAL_OPTS
+    if (avctx->bframebias)
+        x4->params.i_bframe_bias              = avctx->bframebias;
+    if (avctx->deblockalpha)
+        x4->params.i_deblocking_filter_alphac0 = avctx->deblockalpha;
+    if (avctx->deblockbeta)
+        x4->params.i_deblocking_filter_beta    = avctx->deblockbeta;
+    if (avctx->complexityblur >= 0)
+        x4->params.rc.f_complexity_blur        = avctx->complexityblur;
+    if (avctx->directpred >= 0)
+        x4->params.analyse.i_direct_mv_pred    = avctx->directpred;
+    if (avctx->partitions) {
+        if (avctx->partitions & XAVS_PART_I8X8)
+            x4->params.analyse.inter |= XAVS_ANALYSE_I8x8;
+        if (avctx->partitions & XAVS_PART_P8X8)
+            x4->params.analyse.inter |= XAVS_ANALYSE_PSUB16x16;
+        if (avctx->partitions & XAVS_PART_B8X8)
+            x4->params.analyse.inter |= XAVS_ANALYSE_BSUB16x16;
+    }
+    x4->params.rc.b_mb_tree               = !!(avctx->flags2 & CODEC_FLAG2_MBTREE);
+    x4->params.b_aud          = avctx->flags2 & CODEC_FLAG2_AUD;
+    x4->params.analyse.b_mixed_references = avctx->flags2 & CODEC_FLAG2_MIXED_REFS;
+    x4->params.analyse.b_fast_pskip       = avctx->flags2 & CODEC_FLAG2_FASTPSKIP;
+    x4->params.analyse.b_weighted_bipred = avctx->flags2 & CODEC_FLAG2_WPRED;
+#endif
+
+    if (x4->aud >= 0)
+        x4->params.b_aud                      = x4->aud;
+    if (x4->mbtree >= 0)
+        x4->params.rc.b_mb_tree               = x4->mbtree;
+    if (x4->direct_pred >= 0)
+        x4->params.analyse.i_direct_mv_pred   = x4->direct_pred;
+    if (x4->fast_pskip >= 0)
+        x4->params.analyse.b_fast_pskip       = x4->fast_pskip;
+    if (x4->mixed_refs >= 0)
+        x4->params.analyse.b_mixed_references = x4->mixed_refs;
+    if (x4->b_bias != INT_MIN)
+        x4->params.i_bframe_bias              = x4->b_bias;
+    if (x4->cplxblur >= 0)
+        x4->params.rc.f_complexity_blur = x4->cplxblur;
 
     x4->params.i_bframe          = avctx->max_b_frames;
     /* cabac is not included in AVS JiZhun Profile */
     x4->params.b_cabac           = 0;
 
     x4->params.i_bframe_adaptive = avctx->b_frame_strategy;
-    x4->params.i_bframe_bias     = avctx->bframebias;
 
     avctx->has_b_frames          = !!avctx->max_b_frames;
 
@@ -220,8 +280,6 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
     x4->params.i_scenecut_threshold        = avctx->scenechange_threshold;
 
    // x4->params.b_deblocking_filter       = avctx->flags & CODEC_FLAG_LOOP_FILTER;
-    x4->params.i_deblocking_filter_alphac0 = avctx->deblockalpha;
-    x4->params.i_deblocking_filter_beta    = avctx->deblockbeta;
 
     x4->params.rc.i_qp_min                 = avctx->qmin;
     x4->params.rc.i_qp_max                 = avctx->qmax;
@@ -229,7 +287,6 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
 
     x4->params.rc.f_qcompress       = avctx->qcompress; /* 0.0 => cbr, 1.0 => constant qp */
     x4->params.rc.f_qblur           = avctx->qblur;     /* temporally blur quants */
-    x4->params.rc.f_complexity_blur = avctx->complexityblur;
 
     x4->params.i_frame_reference    = avctx->refs;
 
@@ -241,20 +298,6 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
     x4->params.i_fps_num            = avctx->time_base.den;
     x4->params.i_fps_den            = avctx->time_base.num;
     x4->params.analyse.inter        = XAVS_ANALYSE_I8x8 |XAVS_ANALYSE_PSUB16x16| XAVS_ANALYSE_BSUB16x16;
-    if (avctx->partitions) {
-        if (avctx->partitions & XAVS_PART_I8X8)
-            x4->params.analyse.inter |= XAVS_ANALYSE_I8x8;
-
-        if (avctx->partitions & XAVS_PART_P8X8)
-            x4->params.analyse.inter |= XAVS_ANALYSE_PSUB16x16;
-
-        if (avctx->partitions & XAVS_PART_B8X8)
-            x4->params.analyse.inter |= XAVS_ANALYSE_BSUB16x16;
-    }
-
-    x4->params.analyse.i_direct_mv_pred  = avctx->directpred;
-
-    x4->params.analyse.b_weighted_bipred = avctx->flags2 & CODEC_FLAG2_WPRED;
 
     switch (avctx->me_method) {
          case  ME_EPZS:
@@ -279,11 +322,9 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
     x4->params.analyse.i_me_range = avctx->me_range;
     x4->params.analyse.i_subpel_refine    = avctx->me_subpel_quality;
 
-    x4->params.analyse.b_mixed_references = avctx->flags2 & CODEC_FLAG2_MIXED_REFS;
     x4->params.analyse.b_chroma_me        = avctx->me_cmp & FF_CMP_CHROMA;
     /* AVS P2 only enables 8x8 transform */
     x4->params.analyse.b_transform_8x8    = 1; //avctx->flags2 & CODEC_FLAG2_8X8DCT;
-    x4->params.analyse.b_fast_pskip       = avctx->flags2 & CODEC_FLAG2_FASTPSKIP;
 
     x4->params.analyse.i_trellis          = avctx->trellis;
     x4->params.analyse.i_noise_reduction  = avctx->noise_reduction;
@@ -303,14 +344,12 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
 
     /* TAG:do we have MB tree RC method */
     /* what is the RC method we are now using? Default NO */
-    x4->params.rc.b_mb_tree               = !!(avctx->flags2 & CODEC_FLAG2_MBTREE);
     x4->params.rc.f_ip_factor             = 1 / fabs(avctx->i_quant_factor);
     x4->params.rc.f_pb_factor             = avctx->b_quant_factor;
     x4->params.analyse.i_chroma_qp_offset = avctx->chromaoffset;
 
     x4->params.analyse.b_psnr = avctx->flags & CODEC_FLAG_PSNR;
     x4->params.i_log_level    = XAVS_LOG_DEBUG;
-    x4->params.b_aud          = avctx->flags2 & CODEC_FLAG2_AUD;
     x4->params.i_threads      = avctx->thread_count;
     x4->params.b_interlaced   = avctx->flags & CODEC_FLAG_INTERLACED_DCT;
 
@@ -336,6 +375,37 @@ static av_cold int XAVS_init(AVCodecContext *avctx)
     return 0;
 }
 
+#define OFFSET(x) offsetof(XavsContext, x)
+#define VE AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
+static const AVOption options[] = {
+    { "crf",           "Select the quality for constant quality mode",    OFFSET(crf),           FF_OPT_TYPE_FLOAT,  {-1 }, -1, FLT_MAX, VE },
+    { "qp",            "Constant quantization parameter rate control method",OFFSET(cqp),        FF_OPT_TYPE_INT,    {-1 }, -1, INT_MAX, VE },
+    { "b-bias",        "Influences how often B-frames are used",          OFFSET(b_bias),        FF_OPT_TYPE_INT,    {INT_MIN}, INT_MIN, INT_MAX, VE },
+    { "cplxblur",      "Reduce fluctuations in QP (before curve compression)", OFFSET(cplxblur), FF_OPT_TYPE_FLOAT,  {-1 }, -1, FLT_MAX, VE},
+    { "direct-pred",   "Direct MV prediction mode",                       OFFSET(direct_pred),   FF_OPT_TYPE_INT,    {-1 }, -1, INT_MAX, VE, "direct-pred" },
+    { "none",          NULL,      0,    FF_OPT_TYPE_CONST, { XAVS_DIRECT_PRED_NONE },     0, 0, VE, "direct-pred" },
+    { "spatial",       NULL,      0,    FF_OPT_TYPE_CONST, { XAVS_DIRECT_PRED_SPATIAL },  0, 0, VE, "direct-pred" },
+    { "temporal",      NULL,      0,    FF_OPT_TYPE_CONST, { XAVS_DIRECT_PRED_TEMPORAL }, 0, 0, VE, "direct-pred" },
+    { "auto",          NULL,      0,    FF_OPT_TYPE_CONST, { XAVS_DIRECT_PRED_AUTO },     0, 0, VE, "direct-pred" },
+    { "aud",           "Use access unit delimiters.",                     OFFSET(aud),           FF_OPT_TYPE_INT,    {-1 }, -1, 1, VE},
+    { "mbtree",        "Use macroblock tree ratecontrol.",                OFFSET(mbtree),        FF_OPT_TYPE_INT,    {-1 }, -1, 1, VE},
+    { "mixed-refs",    "One reference per partition, as opposed to one reference per macroblock", OFFSET(mixed_refs), FF_OPT_TYPE_INT, {-1}, -1, 1, VE },
+    { "fast-pskip",    NULL,                                              OFFSET(fast_pskip),    FF_OPT_TYPE_INT,    {-1 }, -1, 1, VE},
+    { NULL },
+};
+
+static const AVClass class = {
+    .class_name = "libxavs",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+static const AVCodecDefault xavs_defaults[] = {
+    { "b",                "0" },
+    { NULL },
+};
+
 AVCodec ff_libxavs_encoder = {
     .name           = "libxavs",
     .type           = AVMEDIA_TYPE_VIDEO,
@@ -347,5 +417,7 @@ AVCodec ff_libxavs_encoder = {
     .capabilities   = CODEC_CAP_DELAY,
     .pix_fmts       = (const enum PixelFormat[]) { PIX_FMT_YUV420P, PIX_FMT_NONE },
     .long_name      = NULL_IF_CONFIG_SMALL("libxavs - the Chinese Audio Video Standard Encoder"),
+    .priv_class     = &class,
+    .defaults       = xavs_defaults,
 };
 
diff --git a/libavcodec/rv34.c b/libavcodec/rv34.c
index 2af66a5..6c51186 100644
--- a/libavcodec/rv34.c
+++ b/libavcodec/rv34.c
@@ -1336,6 +1336,13 @@ static int rv34_decode_slice(RV34DecContext *r, int end, const uint8_t* buf, int
             }
         }
         s->mb_x = s->mb_y = 0;
+    } else {
+        int slice_type = r->si.type ? r->si.type : AV_PICTURE_TYPE_I;
+
+        if (slice_type != s->pict_type) {
+            av_log(s->avctx, AV_LOG_ERROR, "Slice type mismatch\n");
+            return AVERROR_INVALIDDATA;
+        }
     }
 
     r->si.end = end;