text: /*
   * Copyright (c) 2001-2003 The FFmpeg project
   *
   * first version by Francois Revol (revol@free.fr)
   * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
   *   by Mike Melanson (melanson@pcisys.net)
   *
   * 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
   * 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 + AV_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;
      }
  
      return 0;
  error:
      adpcm_encode_close(avctx);
      return ret;
  }
  
  static av_cold int adpcm_encode_close(AVCodecContext *avctx)
  {
      ADPCMEncodeContext *s = avctx->priv_data;
      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_intp2(nibble, 3) & 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, 24576);
  
      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*(unsigned)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, 24576));
  #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)) < 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_ARRAY_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);
  
          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][0], buf, status,
                                         64, 1);
                  for (i = 0; i < 64; i++)
                      put_bits(&pb, 4, buf[i ^ 1]);
                  status->prev_sample = status->predictor;
              } 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);
  
          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_uintp2(c->status[i].step_index, 6);
              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_,                               \
      .long_name      = NULL_IF_CONFIG_SMALL(long_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_,                         \
  }
  
  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");
  