diff options
| author | Tim Redfern <tim@eclectronics.org> | 2013-09-05 17:55:35 +0100 |
|---|---|---|
| committer | Tim Redfern <tim@eclectronics.org> | 2013-09-05 17:55:35 +0100 |
| commit | 741fb4b9e135cfb161a749db88713229038577bb (patch) | |
| tree | 08bc9925659cbcac45162bacf31dc6336d4f60b4 /ffmpeg1/libavcodec/adpcmenc.c | |
| parent | a2e1bf3495b7bfefdaedb8fc737e969ab06df079 (diff) | |
making act segmenter
Diffstat (limited to 'ffmpeg1/libavcodec/adpcmenc.c')
| -rw-r--r-- | ffmpeg1/libavcodec/adpcmenc.c | 725 |
1 files changed, 0 insertions, 725 deletions
diff --git a/ffmpeg1/libavcodec/adpcmenc.c b/ffmpeg1/libavcodec/adpcmenc.c deleted file mode 100644 index 762cf67..0000000 --- a/ffmpeg1/libavcodec/adpcmenc.c +++ /dev/null @@ -1,725 +0,0 @@ -/* - * 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"); |
