/*
  * 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 "put_bits.h"
 #include "bytestream.h"
 #include "adpcm.h"
 #include "adpcm_data.h"

 #include "internal.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)
  *

  * See ADPCM decoder reference documents for codec information.

  */
 
 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_close(AVCodecContext *avctx);
 

 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
 {
     ADPCMEncodeContext *s = avctx->priv_data;
     uint8_t *extradata;
     int i;

     int ret = AVERROR(ENOMEM);
 
     if (avctx->channels > 2) {
         av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
         return AVERROR(EINVAL);
     }

     if (avctx->trellis && (unsigned)avctx->trellis > 16U) {

         av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");

         return AVERROR(EINVAL);

     }
 
     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);

     }
 

     avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
 

     switch (avctx->codec->id) {

     case AV_CODEC_ID_ADPCM_IMA_WAV:

         /* each 16 bits sample gives one nibble
            and we have 4 bytes per channel overhead */
         avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
                             (4 * avctx->channels) + 1;
         /* seems frame_size isn't taken into account...
            have to buffer the samples :-( */

         avctx->block_align = BLKSIZE;

         avctx->bits_per_coded_sample = 4;

         break;

     case AV_CODEC_ID_ADPCM_IMA_QT:

         avctx->frame_size  = 64;

         avctx->block_align = 34 * avctx->channels;
         break;

     case AV_CODEC_ID_ADPCM_MS:

         /* each 16 bits sample gives one nibble
            and we have 7 bytes per channel overhead */

         avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;

         avctx->bits_per_coded_sample = 4;

         avctx->block_align    = BLKSIZE;

         if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
             goto error;

         avctx->extradata_size = 32;

         extradata = avctx->extradata;

         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 AV_CODEC_ID_ADPCM_YAMAHA:

         avctx->frame_size  = BLKSIZE * 2 / avctx->channels;

         avctx->block_align = BLKSIZE;
         break;

     case AV_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");

             ret = AVERROR(EINVAL);

             goto error;
         }
         avctx->frame_size = 512 * (avctx->sample_rate / 11025);
         break;
     default:

         ret = AVERROR(EINVAL);

         goto error;
     }
 

 #if FF_API_OLD_ENCODE_AUDIO

     if (!(avctx->coded_frame = avcodec_alloc_frame()))

         goto error;

 #endif

 
     return 0;
 error:

     adpcm_encode_close(avctx);

     return ret;

 }
 
 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
 {
     ADPCMEncodeContext *s = avctx->priv_data;

 #if FF_API_OLD_ENCODE_AUDIO

     av_freep(&avctx->coded_frame);

 #endif

     av_freep(&s->paths);
     av_freep(&s->node_buf);
     av_freep(&s->nodep_buf);
     av_freep(&s->trellis_hash);
 
     return 0;
 }
 
 

 static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
                                                 int16_t 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 uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
                                                    int16_t 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 uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
                                                int16_t 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 += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;

 
     c->sample2 = c->sample1;
     c->sample1 = av_clip_int16(predictor);
 

     c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;

     if (c->idelta < 16)
         c->idelta = 16;

 
     return nibble;
 }
 

 static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
                                                    int16_t 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 int16_t *samples, uint8_t *dst,

                                    ADPCMChannelStatus *c, int n, int stride)

 {
     //FIXME 6% faster if frontier is a compile-time constant
     ADPCMEncodeContext *s = avctx->priv_data;
     const int frontier = 1 << avctx->trellis;

     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 == AV_CODEC_ID_ADPCM_IMA_WAV ||
         version == AV_CODEC_ID_ADPCM_IMA_QT  ||
         version == AV_CODEC_ID_ADPCM_SWF)

         nodes[0]->sample1 = c->prev_sample;

     if (version == AV_CODEC_ID_ADPCM_MS)

         nodes[0]->step = c->idelta;

     if (version == AV_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 == AV_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) {\

                         av_assert1(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 == AV_CODEC_ID_ADPCM_IMA_WAV ||
                        version == AV_CODEC_ID_ADPCM_IMA_QT  ||
                        version == AV_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 { //AV_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, AVPacket *avpkt,
                               const AVFrame *frame, int *got_packet_ptr)

 {

     int n, i, ch, st, pkt_size, ret;

     const int16_t *samples;

     int16_t **samples_p;

     uint8_t *dst;

     ADPCMEncodeContext *c = avctx->priv_data;
     uint8_t *buf;
 

     samples = (const int16_t *)frame->data[0];

     samples_p = (int16_t **)frame->extended_data;

     st = avctx->channels == 2;

     if (avctx->codec_id == AV_CODEC_ID_ADPCM_SWF)

         pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
     else
         pkt_size = avctx->block_align;

     if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)) < 0)

         return ret;
     dst = avpkt->data;

 
     switch(avctx->codec->id) {

     case AV_CODEC_ID_ADPCM_IMA_WAV:

     {

         int blocks, j;

 
         blocks = (frame->nb_samples - 1) / 8;
 
         for (ch = 0; ch < avctx->channels; ch++) {
             ADPCMChannelStatus *status = &c->status[ch];

             status->prev_sample = samples_p[ch][0];

             /* status->step_index = 0;
                XXX: not sure how to init the state machine */
             bytestream_put_le16(&dst, status->prev_sample);
             *dst++ = status->step_index;
             *dst++ = 0; /* unknown */

         }

         /* stereo: 4 bytes (8 samples) for left, 4 bytes for right */

         if (avctx->trellis > 0) {

             FF_ALLOC_OR_GOTO(avctx, buf, avctx->channels * blocks * 8, error);
             for (ch = 0; ch < avctx->channels; ch++) {

                 adpcm_compress_trellis(avctx, &samples_p[ch][1],

                                        buf + ch * blocks * 8, &c->status[ch],

                                        blocks * 8, 1);

             }
             for (i = 0; i < blocks; i++) {
                 for (ch = 0; ch < avctx->channels; ch++) {
                     uint8_t *buf1 = buf + ch * blocks * 8 + i * 8;
                     for (j = 0; j < 8; j += 2)
                         *dst++ = buf1[j] | (buf1[j + 1] << 4);

                 }

             }
             av_free(buf);
         } else {

             for (i = 0; i < blocks; i++) {
                 for (ch = 0; ch < avctx->channels; ch++) {
                     ADPCMChannelStatus *status = &c->status[ch];

                     const int16_t *smp = &samples_p[ch][1 + i * 8];

                     for (j = 0; j < 8; j += 2) {

                         uint8_t v = adpcm_ima_compress_sample(status, smp[j    ]);
                         v        |= adpcm_ima_compress_sample(status, smp[j + 1]) << 4;

                         *dst++ = v;

                     }

                 }
             }

         }

         break;

     }

     case AV_CODEC_ID_ADPCM_IMA_QT:

     {
         PutBitContext pb;

         init_put_bits(&pb, dst, pkt_size * 8);

         for (ch = 0; ch < avctx->channels; ch++) {

             ADPCMChannelStatus *status = &c->status[ch];
             put_bits(&pb, 9, (status->prev_sample & 0xFFFF) >> 7);
             put_bits(&pb, 7,  status->step_index);

             if (avctx->trellis > 0) {

                 uint8_t buf[64];

                 adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
                                        64, 1);

                 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(status, samples_p[ch][i    ]);
                     t2 = adpcm_ima_qt_compress_sample(status, samples_p[ch][i + 1]);

                     put_bits(&pb, 4, t2);
                     put_bits(&pb, 4, t1);
                 }
             }
         }
 
         flush_put_bits(&pb);
         break;
     }

     case AV_CODEC_ID_ADPCM_SWF:

     {
         PutBitContext pb;

         init_put_bits(&pb, dst, pkt_size * 8);

         n = frame->nb_samples - 1;

         // store AdpcmCodeSize
         put_bits(&pb, 2, 2);    // set 4-bit flash adpcm format

         // init the encoder state
         for (i = 0; i < avctx->channels; i++) {
             // clip step so it fits 6 bits
             c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);

             put_sbits(&pb, 16, samples[i]);
             put_bits(&pb, 6, c->status[i].step_index);

             c->status[i].prev_sample = samples[i];

         }
 

         if (avctx->trellis > 0) {
             FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);

             adpcm_compress_trellis(avctx, samples + avctx->channels, buf,

                                    &c->status[0], n, avctx->channels);

             if (avctx->channels == 2)

                 adpcm_compress_trellis(avctx, samples + avctx->channels + 1,

                                        buf + n, &c->status[1], n,
                                        avctx->channels);

             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 < frame->nb_samples; 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);
         break;
     }

     case AV_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) {

             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,
                                        avctx->channels);

                 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, avctx->channels);
                 adpcm_compress_trellis(avctx, samples + 1, buf + n,
                                        &c->status[1], n, avctx->channels);

                 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 AV_CODEC_ID_ADPCM_YAMAHA:

         n = frame->nb_samples / 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,
                                        avctx->channels);

                 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, avctx->channels);
                 adpcm_compress_trellis(avctx, samples + 1, buf + n,
                                        &c->status[1], n, avctx->channels);

                 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:

         return AVERROR(EINVAL);

     }

 
     avpkt->size = pkt_size;
     *got_packet_ptr = 1;
     return 0;

 error:
     return AVERROR(ENOMEM);

 }
 

 static const enum AVSampleFormat sample_fmts[] = {
     AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
 };

 static const enum AVSampleFormat sample_fmts_p[] = {
     AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE
 };
 
 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \

 AVCodec ff_ ## name_ ## _encoder = {                        \
     .name           = #name_,                               \
     .type           = AVMEDIA_TYPE_AUDIO,                   \
     .id             = id_,                                  \
     .priv_data_size = sizeof(ADPCMEncodeContext),           \
     .init           = adpcm_encode_init,                    \

     .encode2        = adpcm_encode_frame,                   \

     .close          = adpcm_encode_close,                   \

     .sample_fmts    = sample_fmts_,                         \

     .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \

 }
 

 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT,  adpcm_ima_qt,  sample_fmts_p, "ADPCM IMA QuickTime");
 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, sample_fmts_p, "ADPCM IMA WAV");
 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_MS,      adpcm_ms,      sample_fmts,   "ADPCM Microsoft");
 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_SWF,     adpcm_swf,     sample_fmts,   "ADPCM Shockwave Flash");
 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_YAMAHA,  adpcm_yamaha,  sample_fmts,   "ADPCM Yamaha");