adding ffmpeg specific version

1 files changed, 725 insertions, 0 deletions

diff --git a/ffmpeg1/libavcodec/adpcmenc.c b/ffmpeg1/libavcodec/adpcmenc.c
new file mode 100644
index 0000000..762cf67
--- /dev/null
+++ b/ffmpeg1/libavcodec/adpcmenc.c
@@ -0,0 +1,725 @@
+/*
+ * 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;
+    }
+
+    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(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");