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authorJustin Ruggles <justin.ruggles@gmail.com>2011-09-07 18:34:09 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2011-09-12 11:26:11 -0400
commit826c56d16e55f3819a75d01f957dd295aa1e9f3a (patch)
tree65ffaad2a28d590487ca681c343de2b0a79a48b6 /libavcodec/adpcmenc.c
parent57650c70e22b8259f4ac65d5826a667c8f67726e (diff)
adpcm: split ADPCM encoders and decoders into separate files.
Move shared tables to a separate file as well.
Diffstat (limited to 'libavcodec/adpcmenc.c')
-rw-r--r--libavcodec/adpcmenc.c655
1 files changed, 655 insertions, 0 deletions
diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c
new file mode 100644
index 0000000000..ec062849bd
--- /dev/null
+++ b/libavcodec/adpcmenc.c
@@ -0,0 +1,655 @@
+/*
+ * Copyright (c) 2001-2003 The ffmpeg Project
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "bytestream.h"
+#include "adpcm.h"
+#include "adpcm_data.h"
+
+/**
+ * @file
+ * ADPCM encoders
+ * First version by Francois Revol (revol@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;
+ /* seems frame_size isn't taken into account... have to buffer the samples :-( */
+ break;
+ case CODEC_ID_ADPCM_IMA_QT:
+ avctx->frame_size = 64;
+ avctx->block_align = 34 * avctx->channels;
+ break;
+ case CODEC_ID_ADPCM_MS:
+ avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */
+ /* and we have 7 bytes per channel overhead */
+ avctx->block_align = BLKSIZE;
+ avctx->extradata_size = 32;
+ extradata = avctx->extradata = av_malloc(avctx->extradata_size);
+ if (!extradata)
+ return AVERROR(ENOMEM);
+ bytestream_put_le16(&extradata, avctx->frame_size);
+ bytestream_put_le16(&extradata, 7); /* wNumCoef */
+ for (i = 0; i < 7; i++) {
+ bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
+ bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
+ }
+ break;
+ case CODEC_ID_ADPCM_YAMAHA:
+ avctx->frame_size = BLKSIZE * avctx->channels;
+ avctx->block_align = BLKSIZE;
+ break;
+ case CODEC_ID_ADPCM_SWF:
+ if (avctx->sample_rate != 11025 &&
+ avctx->sample_rate != 22050 &&
+ avctx->sample_rate != 44100) {
+ av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, 22050 or 44100\n");
+ goto error;
+ }
+ avctx->frame_size = 512 * (avctx->sample_rate / 11025);
+ break;
+ default:
+ goto error;
+ }
+
+ avctx->coded_frame= avcodec_alloc_frame();
+ avctx->coded_frame->key_frame= 1;
+
+ return 0;
+error:
+ av_freep(&s->paths);
+ av_freep(&s->node_buf);
+ av_freep(&s->nodep_buf);
+ av_freep(&s->trellis_hash);
+ return -1;
+}
+
+static av_cold int adpcm_encode_close(AVCodecContext *avctx)
+{
+ ADPCMEncodeContext *s = avctx->priv_data;
+ av_freep(&avctx->coded_frame);
+ av_freep(&s->paths);
+ av_freep(&s->node_buf);
+ av_freep(&s->nodep_buf);
+ av_freep(&s->trellis_hash);
+
+ return 0;
+}
+
+
+static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, short sample)
+{
+ int delta = sample - c->prev_sample;
+ int nibble = FFMIN(7, abs(delta)*4/ff_adpcm_step_table[c->step_index]) + (delta<0)*8;
+ c->prev_sample += ((ff_adpcm_step_table[c->step_index] * ff_adpcm_yamaha_difflookup[nibble]) / 8);
+ c->prev_sample = av_clip_int16(c->prev_sample);
+ c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
+ return nibble;
+}
+
+static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, short sample)
+{
+ int predictor, nibble, bias;
+
+ predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
+
+ nibble= sample - predictor;
+ if(nibble>=0) bias= c->idelta/2;
+ else bias=-c->idelta/2;
+
+ nibble= (nibble + bias) / c->idelta;
+ nibble= av_clip(nibble, -8, 7)&0x0F;
+
+ predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
+
+ c->sample2 = c->sample1;
+ c->sample1 = av_clip_int16(predictor);
+
+ c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
+ if (c->idelta < 16) c->idelta = 16;
+
+ return nibble;
+}
+
+static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, short sample)
+{
+ int nibble, delta;
+
+ if(!c->step) {
+ c->predictor = 0;
+ c->step = 127;
+ }
+
+ delta = sample - c->predictor;
+
+ nibble = FFMIN(7, abs(delta)*4/c->step) + (delta<0)*8;
+
+ c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
+ c->predictor = av_clip_int16(c->predictor);
+ c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
+ c->step = av_clip(c->step, 127, 24567);
+
+ return nibble;
+}
+
+static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
+ uint8_t *dst, ADPCMChannelStatus *c, int n)
+{
+ //FIXME 6% faster if frontier is a compile-time constant
+ ADPCMEncodeContext *s = avctx->priv_data;
+ const int frontier = 1 << avctx->trellis;
+ const int stride = avctx->channels;
+ const int version = avctx->codec->id;
+ TrellisPath *paths = s->paths, *p;
+ TrellisNode *node_buf = s->node_buf;
+ TrellisNode **nodep_buf = s->nodep_buf;
+ TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
+ TrellisNode **nodes_next = nodep_buf + frontier;
+ int pathn = 0, froze = -1, i, j, k, generation = 0;
+ uint8_t *hash = s->trellis_hash;
+ memset(hash, 0xff, 65536 * sizeof(*hash));
+
+ memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
+ nodes[0] = node_buf + frontier;
+ nodes[0]->ssd = 0;
+ nodes[0]->path = 0;
+ nodes[0]->step = c->step_index;
+ nodes[0]->sample1 = c->sample1;
+ nodes[0]->sample2 = c->sample2;
+ if((version == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_IMA_QT) || (version == CODEC_ID_ADPCM_SWF))
+ nodes[0]->sample1 = c->prev_sample;
+ if(version == CODEC_ID_ADPCM_MS)
+ nodes[0]->step = c->idelta;
+ if(version == CODEC_ID_ADPCM_YAMAHA) {
+ if(c->step == 0) {
+ nodes[0]->step = 127;
+ nodes[0]->sample1 = 0;
+ } else {
+ nodes[0]->step = c->step;
+ nodes[0]->sample1 = c->predictor;
+ }
+ }
+
+ for(i=0; i<n; i++) {
+ TrellisNode *t = node_buf + frontier*(i&1);
+ TrellisNode **u;
+ int sample = samples[i*stride];
+ int heap_pos = 0;
+ memset(nodes_next, 0, frontier*sizeof(TrellisNode*));
+ for(j=0; j<frontier && nodes[j]; j++) {
+ // higher j have higher ssd already, so they're likely to yield a suboptimal next sample too
+ const int range = (j < frontier/2) ? 1 : 0;
+ const int step = nodes[j]->step;
+ int nidx;
+ if(version == CODEC_ID_ADPCM_MS) {
+ const int predictor = ((nodes[j]->sample1 * c->coeff1) + (nodes[j]->sample2 * c->coeff2)) / 64;
+ const int div = (sample - predictor) / step;
+ const int nmin = av_clip(div-range, -8, 6);
+ const int nmax = av_clip(div+range, -7, 7);
+ for(nidx=nmin; nidx<=nmax; nidx++) {
+ const int nibble = nidx & 0xf;
+ int dec_sample = predictor + nidx * step;
+#define STORE_NODE(NAME, STEP_INDEX)\
+ int d;\
+ uint32_t ssd;\
+ int pos;\
+ TrellisNode *u;\
+ uint8_t *h;\
+ dec_sample = av_clip_int16(dec_sample);\
+ d = sample - dec_sample;\
+ ssd = nodes[j]->ssd + d*d;\
+ /* Check for wraparound, skip such samples completely. \
+ * Note, changing ssd to a 64 bit variable would be \
+ * simpler, avoiding this check, but it's slower on \
+ * x86 32 bit at the moment. */\
+ if (ssd < nodes[j]->ssd)\
+ goto next_##NAME;\
+ /* Collapse any two states with the same previous sample value. \
+ * One could also distinguish states by step and by 2nd to last
+ * sample, but the effects of that are negligible.
+ * Since nodes in the previous generation are iterated
+ * through a heap, they're roughly ordered from better to
+ * worse, but not strictly ordered. Therefore, an earlier
+ * node with the same sample value is better in most cases
+ * (and thus the current is skipped), but not strictly
+ * in all cases. Only skipping samples where ssd >=
+ * ssd of the earlier node with the same sample gives
+ * slightly worse quality, though, for some reason. */ \
+ h = &hash[(uint16_t) dec_sample];\
+ if (*h == generation)\
+ goto next_##NAME;\
+ if (heap_pos < frontier) {\
+ pos = heap_pos++;\
+ } else {\
+ /* Try to replace one of the leaf nodes with the new \
+ * one, but try a different slot each time. */\
+ pos = (frontier >> 1) + (heap_pos & ((frontier >> 1) - 1));\
+ if (ssd > nodes_next[pos]->ssd)\
+ goto next_##NAME;\
+ heap_pos++;\
+ }\
+ *h = generation;\
+ u = nodes_next[pos];\
+ if(!u) {\
+ assert(pathn < FREEZE_INTERVAL<<avctx->trellis);\
+ u = t++;\
+ nodes_next[pos] = u;\
+ u->path = pathn++;\
+ }\
+ u->ssd = ssd;\
+ u->step = STEP_INDEX;\
+ u->sample2 = nodes[j]->sample1;\
+ u->sample1 = dec_sample;\
+ paths[u->path].nibble = nibble;\
+ paths[u->path].prev = nodes[j]->path;\
+ /* Sift the newly inserted node up in the heap to \
+ * restore the heap property. */\
+ while (pos > 0) {\
+ int parent = (pos - 1) >> 1;\
+ if (nodes_next[parent]->ssd <= ssd)\
+ break;\
+ FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
+ pos = parent;\
+ }\
+ next_##NAME:;
+ STORE_NODE(ms, FFMAX(16, (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
+ }
+ } else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (version == CODEC_ID_ADPCM_IMA_QT)|| (version == CODEC_ID_ADPCM_SWF)) {
+#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
+ const int predictor = nodes[j]->sample1;\
+ const int div = (sample - predictor) * 4 / STEP_TABLE;\
+ int nmin = av_clip(div-range, -7, 6);\
+ int nmax = av_clip(div+range, -6, 7);\
+ if(nmin<=0) nmin--; /* distinguish -0 from +0 */\
+ if(nmax<0) nmax--;\
+ for(nidx=nmin; nidx<=nmax; nidx++) {\
+ const int nibble = nidx<0 ? 7-nidx : nidx;\
+ int dec_sample = predictor + (STEP_TABLE * ff_adpcm_yamaha_difflookup[nibble]) / 8;\
+ STORE_NODE(NAME, STEP_INDEX);\
+ }
+ LOOP_NODES(ima, ff_adpcm_step_table[step], av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
+ } else { //CODEC_ID_ADPCM_YAMAHA
+ LOOP_NODES(yamaha, step, av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8, 127, 24567));
+#undef LOOP_NODES
+#undef STORE_NODE
+ }
+ }
+
+ u = nodes;
+ nodes = nodes_next;
+ nodes_next = u;
+
+ generation++;
+ if (generation == 255) {
+ memset(hash, 0xff, 65536 * sizeof(*hash));
+ generation = 0;
+ }
+
+ // prevent overflow
+ if(nodes[0]->ssd > (1<<28)) {
+ for(j=1; j<frontier && nodes[j]; j++)
+ nodes[j]->ssd -= nodes[0]->ssd;
+ nodes[0]->ssd = 0;
+ }
+
+ // merge old paths to save memory
+ if(i == froze + FREEZE_INTERVAL) {
+ p = &paths[nodes[0]->path];
+ for(k=i; k>froze; k--) {
+ dst[k] = p->nibble;
+ p = &paths[p->prev];
+ }
+ froze = i;
+ pathn = 0;
+ // other nodes might use paths that don't coincide with the frozen one.
+ // checking which nodes do so is too slow, so just kill them all.
+ // this also slightly improves quality, but I don't know why.
+ memset(nodes+1, 0, (frontier-1)*sizeof(TrellisNode*));
+ }
+ }
+
+ p = &paths[nodes[0]->path];
+ for(i=n-1; i>froze; i--) {
+ dst[i] = p->nibble;
+ p = &paths[p->prev];
+ }
+
+ c->predictor = nodes[0]->sample1;
+ c->sample1 = nodes[0]->sample1;
+ c->sample2 = nodes[0]->sample2;
+ c->step_index = nodes[0]->step;
+ c->step = nodes[0]->step;
+ c->idelta = nodes[0]->step;
+}
+
+static int adpcm_encode_frame(AVCodecContext *avctx,
+ unsigned char *frame, int buf_size, void *data)
+{
+ int n, i, st;
+ short *samples;
+ unsigned char *dst;
+ ADPCMEncodeContext *c = avctx->priv_data;
+ uint8_t *buf;
+
+ dst = frame;
+ samples = (short *)data;
+ st= avctx->channels == 2;
+/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
+
+ switch(avctx->codec->id) {
+ case CODEC_ID_ADPCM_IMA_WAV:
+ n = avctx->frame_size / 8;
+ c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
+/* c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */
+ bytestream_put_le16(&dst, c->status[0].prev_sample);
+ *dst++ = (unsigned char)c->status[0].step_index;
+ *dst++ = 0; /* unknown */
+ samples++;
+ if (avctx->channels == 2) {
+ c->status[1].prev_sample = (signed short)samples[0];
+/* c->status[1].step_index = 0; */
+ bytestream_put_le16(&dst, c->status[1].prev_sample);
+ *dst++ = (unsigned char)c->status[1].step_index;
+ *dst++ = 0;
+ samples++;
+ }
+
+ /* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */
+ if(avctx->trellis > 0) {
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error);
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8);
+ if(avctx->channels == 2)
+ adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8);
+ for(i=0; i<n; i++) {
+ *dst++ = buf[8*i+0] | (buf[8*i+1] << 4);
+ *dst++ = buf[8*i+2] | (buf[8*i+3] << 4);
+ *dst++ = buf[8*i+4] | (buf[8*i+5] << 4);
+ *dst++ = buf[8*i+6] | (buf[8*i+7] << 4);
+ if (avctx->channels == 2) {
+ uint8_t *buf1 = buf + n*8;
+ *dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4);
+ *dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4);
+ *dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4);
+ *dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4);
+ }
+ }
+ av_free(buf);
+ } else
+ for (; n>0; n--) {
+ *dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
+ *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4;
+ dst++;
+ *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
+ *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
+ dst++;
+ *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
+ *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
+ dst++;
+ *dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
+ *dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
+ dst++;
+ /* right channel */
+ if (avctx->channels == 2) {
+ *dst = adpcm_ima_compress_sample(&c->status[1], samples[1]);
+ *dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4;
+ dst++;
+ *dst = adpcm_ima_compress_sample(&c->status[1], samples[5]);
+ *dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4;
+ dst++;
+ *dst = adpcm_ima_compress_sample(&c->status[1], samples[9]);
+ *dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
+ dst++;
+ *dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
+ *dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
+ dst++;
+ }
+ samples += 8 * avctx->channels;
+ }
+ break;
+ case CODEC_ID_ADPCM_IMA_QT:
+ {
+ int ch, i;
+ PutBitContext pb;
+ init_put_bits(&pb, dst, buf_size*8);
+
+ for(ch=0; ch<avctx->channels; ch++){
+ put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
+ put_bits(&pb, 7, c->status[ch].step_index);
+ if(avctx->trellis > 0) {
+ uint8_t buf[64];
+ adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
+ for(i=0; i<64; i++)
+ put_bits(&pb, 4, buf[i^1]);
+ c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F;
+ } else {
+ for (i=0; i<64; i+=2){
+ int t1, t2;
+ t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
+ t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
+ put_bits(&pb, 4, t2);
+ put_bits(&pb, 4, t1);
+ }
+ c->status[ch].prev_sample &= ~0x7F;
+ }
+ }
+
+ flush_put_bits(&pb);
+ dst += put_bits_count(&pb)>>3;
+ break;
+ }
+ case CODEC_ID_ADPCM_SWF:
+ {
+ int i;
+ PutBitContext pb;
+ init_put_bits(&pb, dst, buf_size*8);
+
+ n = avctx->frame_size-1;
+
+ //Store AdpcmCodeSize
+ put_bits(&pb, 2, 2); //Set 4bits flash adpcm format
+
+ //Init the encoder state
+ for(i=0; i<avctx->channels; i++){
+ c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits
+ put_sbits(&pb, 16, samples[i]);
+ put_bits(&pb, 6, c->status[i].step_index);
+ c->status[i].prev_sample = (signed short)samples[i];
+ }
+
+ if(avctx->trellis > 0) {
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
+ adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n);
+ if (avctx->channels == 2)
+ adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n);
+ for(i=0; i<n; i++) {
+ put_bits(&pb, 4, buf[i]);
+ if (avctx->channels == 2)
+ put_bits(&pb, 4, buf[n+i]);
+ }
+ av_free(buf);
+ } else {
+ for (i=1; i<avctx->frame_size; i++) {
+ put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i]));
+ if (avctx->channels == 2)
+ put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1]));
+ }
+ }
+ flush_put_bits(&pb);
+ dst += put_bits_count(&pb)>>3;
+ break;
+ }
+ case CODEC_ID_ADPCM_MS:
+ for(i=0; i<avctx->channels; i++){
+ int predictor=0;
+
+ *dst++ = predictor;
+ c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
+ c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
+ }
+ for(i=0; i<avctx->channels; i++){
+ if (c->status[i].idelta < 16)
+ c->status[i].idelta = 16;
+
+ bytestream_put_le16(&dst, c->status[i].idelta);
+ }
+ for(i=0; i<avctx->channels; i++){
+ c->status[i].sample2= *samples++;
+ }
+ for(i=0; i<avctx->channels; i++){
+ c->status[i].sample1= *samples++;
+
+ bytestream_put_le16(&dst, c->status[i].sample1);
+ }
+ for(i=0; i<avctx->channels; i++)
+ bytestream_put_le16(&dst, c->status[i].sample2);
+
+ if(avctx->trellis > 0) {
+ int n = avctx->block_align - 7*avctx->channels;
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
+ if(avctx->channels == 1) {
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ for(i=0; i<n; i+=2)
+ *dst++ = (buf[i] << 4) | buf[i+1];
+ } else {
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
+ for(i=0; i<n; i++)
+ *dst++ = (buf[i] << 4) | buf[n+i];
+ }
+ av_free(buf);
+ } else
+ for(i=7*avctx->channels; i<avctx->block_align; i++) {
+ int nibble;
+ nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++)<<4;
+ nibble|= adpcm_ms_compress_sample(&c->status[st], *samples++);
+ *dst++ = nibble;
+ }
+ break;
+ case CODEC_ID_ADPCM_YAMAHA:
+ n = avctx->frame_size / 2;
+ if(avctx->trellis > 0) {
+ FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error);
+ n *= 2;
+ if(avctx->channels == 1) {
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ for(i=0; i<n; i+=2)
+ *dst++ = buf[i] | (buf[i+1] << 4);
+ } else {
+ adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
+ adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
+ for(i=0; i<n; i++)
+ *dst++ = buf[i] | (buf[n+i] << 4);
+ }
+ av_free(buf);
+ } else
+ for (n *= avctx->channels; n>0; n--) {
+ int nibble;
+ nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
+ nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
+ *dst++ = nibble;
+ }
+ break;
+ default:
+ error:
+ return -1;
+ }
+ return dst - frame;
+}
+
+
+#define ADPCM_ENCODER(id,name,long_name_) \
+AVCodec ff_ ## name ## _encoder = { \
+ #name, \
+ AVMEDIA_TYPE_AUDIO, \
+ id, \
+ sizeof(ADPCMEncodeContext), \
+ adpcm_encode_init, \
+ adpcm_encode_frame, \
+ adpcm_encode_close, \
+ NULL, \
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \
+ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
+}
+
+ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
+ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
+ADPCM_ENCODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
+ADPCM_ENCODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
+ADPCM_ENCODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");