/* * 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");