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727 lines
27 KiB
C
727 lines
27 KiB
C
/*
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* Copyright (c) 2001-2003 The ffmpeg Project
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*
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* first version by Francois Revol (revol@free.fr)
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* fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
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* by Mike Melanson (melanson@pcisys.net)
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "get_bits.h"
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#include "put_bits.h"
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#include "bytestream.h"
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#include "adpcm.h"
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#include "adpcm_data.h"
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#include "internal.h"
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/**
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* @file
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* ADPCM encoders
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* See ADPCM decoder reference documents for codec information.
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*/
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typedef struct TrellisPath {
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int nibble;
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int prev;
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} TrellisPath;
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typedef struct TrellisNode {
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uint32_t ssd;
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int path;
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int sample1;
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int sample2;
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int step;
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} TrellisNode;
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typedef struct ADPCMEncodeContext {
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ADPCMChannelStatus status[6];
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TrellisPath *paths;
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TrellisNode *node_buf;
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TrellisNode **nodep_buf;
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uint8_t *trellis_hash;
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} ADPCMEncodeContext;
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#define FREEZE_INTERVAL 128
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static av_cold int adpcm_encode_init(AVCodecContext *avctx)
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{
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ADPCMEncodeContext *s = avctx->priv_data;
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uint8_t *extradata;
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int i;
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int ret = AVERROR(ENOMEM);
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if (avctx->channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
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return AVERROR(EINVAL);
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}
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if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
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av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
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return AVERROR(EINVAL);
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}
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if (avctx->trellis) {
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int frontier = 1 << avctx->trellis;
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int max_paths = frontier * FREEZE_INTERVAL;
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FF_ALLOC_OR_GOTO(avctx, s->paths,
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max_paths * sizeof(*s->paths), error);
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FF_ALLOC_OR_GOTO(avctx, s->node_buf,
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2 * frontier * sizeof(*s->node_buf), error);
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FF_ALLOC_OR_GOTO(avctx, s->nodep_buf,
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2 * frontier * sizeof(*s->nodep_buf), error);
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FF_ALLOC_OR_GOTO(avctx, s->trellis_hash,
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65536 * sizeof(*s->trellis_hash), error);
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}
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avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
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switch (avctx->codec->id) {
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case AV_CODEC_ID_ADPCM_IMA_WAV:
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/* each 16 bits sample gives one nibble
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and we have 4 bytes per channel overhead */
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avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 /
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(4 * avctx->channels) + 1;
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/* seems frame_size isn't taken into account...
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have to buffer the samples :-( */
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avctx->block_align = BLKSIZE;
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break;
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case AV_CODEC_ID_ADPCM_IMA_QT:
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avctx->frame_size = 64;
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avctx->block_align = 34 * avctx->channels;
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break;
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case AV_CODEC_ID_ADPCM_MS:
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/* each 16 bits sample gives one nibble
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and we have 7 bytes per channel overhead */
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avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 /
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avctx->channels + 2;
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avctx->block_align = BLKSIZE;
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if (!(avctx->extradata = av_malloc(32 + AV_INPUT_BUFFER_PADDING_SIZE)))
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goto error;
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avctx->extradata_size = 32;
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extradata = avctx->extradata;
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bytestream_put_le16(&extradata, avctx->frame_size);
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bytestream_put_le16(&extradata, 7); /* wNumCoef */
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for (i = 0; i < 7; i++) {
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bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff1[i] * 4);
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bytestream_put_le16(&extradata, ff_adpcm_AdaptCoeff2[i] * 4);
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}
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break;
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case AV_CODEC_ID_ADPCM_YAMAHA:
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avctx->frame_size = BLKSIZE * 2 / avctx->channels;
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avctx->block_align = BLKSIZE;
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break;
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case AV_CODEC_ID_ADPCM_SWF:
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if (avctx->sample_rate != 11025 &&
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avctx->sample_rate != 22050 &&
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avctx->sample_rate != 44100) {
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av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
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"22050 or 44100\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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avctx->frame_size = 512 * (avctx->sample_rate / 11025);
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break;
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default:
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ret = AVERROR(EINVAL);
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goto error;
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}
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return 0;
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error:
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av_freep(&s->paths);
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av_freep(&s->node_buf);
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av_freep(&s->nodep_buf);
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av_freep(&s->trellis_hash);
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return ret;
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}
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static av_cold int adpcm_encode_close(AVCodecContext *avctx)
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{
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ADPCMEncodeContext *s = avctx->priv_data;
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av_freep(&s->paths);
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av_freep(&s->node_buf);
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av_freep(&s->nodep_buf);
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av_freep(&s->trellis_hash);
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return 0;
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}
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static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
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int16_t sample)
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{
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int delta = sample - c->prev_sample;
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int nibble = FFMIN(7, abs(delta) * 4 /
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ff_adpcm_step_table[c->step_index]) + (delta < 0) * 8;
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c->prev_sample += ((ff_adpcm_step_table[c->step_index] *
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ff_adpcm_yamaha_difflookup[nibble]) / 8);
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c->prev_sample = av_clip_int16(c->prev_sample);
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c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
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return nibble;
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}
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static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
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int16_t sample)
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{
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int delta = sample - c->prev_sample;
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int mask, step = ff_adpcm_step_table[c->step_index];
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int diff = step >> 3;
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int nibble = 0;
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if (delta < 0) {
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nibble = 8;
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delta = -delta;
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}
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for (mask = 4; mask;) {
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if (delta >= step) {
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nibble |= mask;
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delta -= step;
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diff += step;
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}
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step >>= 1;
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mask >>= 1;
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}
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if (nibble & 8)
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c->prev_sample -= diff;
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else
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c->prev_sample += diff;
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c->prev_sample = av_clip_int16(c->prev_sample);
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c->step_index = av_clip(c->step_index + ff_adpcm_index_table[nibble], 0, 88);
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return nibble;
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}
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static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
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int16_t sample)
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{
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int predictor, nibble, bias;
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predictor = (((c->sample1) * (c->coeff1)) +
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(( c->sample2) * (c->coeff2))) / 64;
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nibble = sample - predictor;
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if (nibble >= 0)
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bias = c->idelta / 2;
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else
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bias = -c->idelta / 2;
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nibble = (nibble + bias) / c->idelta;
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nibble = av_clip(nibble, -8, 7) & 0x0F;
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predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
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c->sample2 = c->sample1;
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c->sample1 = av_clip_int16(predictor);
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c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
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if (c->idelta < 16)
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c->idelta = 16;
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return nibble;
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}
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static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
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int16_t sample)
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{
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int nibble, delta;
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if (!c->step) {
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c->predictor = 0;
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c->step = 127;
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}
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delta = sample - c->predictor;
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nibble = FFMIN(7, abs(delta) * 4 / c->step) + (delta < 0) * 8;
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c->predictor += ((c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8);
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c->predictor = av_clip_int16(c->predictor);
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c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
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c->step = av_clip(c->step, 127, 24567);
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return nibble;
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}
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static void adpcm_compress_trellis(AVCodecContext *avctx,
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const int16_t *samples, uint8_t *dst,
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ADPCMChannelStatus *c, int n, int stride)
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{
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//FIXME 6% faster if frontier is a compile-time constant
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ADPCMEncodeContext *s = avctx->priv_data;
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const int frontier = 1 << avctx->trellis;
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const int version = avctx->codec->id;
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TrellisPath *paths = s->paths, *p;
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TrellisNode *node_buf = s->node_buf;
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TrellisNode **nodep_buf = s->nodep_buf;
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TrellisNode **nodes = nodep_buf; // nodes[] is always sorted by .ssd
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TrellisNode **nodes_next = nodep_buf + frontier;
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int pathn = 0, froze = -1, i, j, k, generation = 0;
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uint8_t *hash = s->trellis_hash;
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memset(hash, 0xff, 65536 * sizeof(*hash));
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memset(nodep_buf, 0, 2 * frontier * sizeof(*nodep_buf));
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nodes[0] = node_buf + frontier;
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nodes[0]->ssd = 0;
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nodes[0]->path = 0;
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nodes[0]->step = c->step_index;
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nodes[0]->sample1 = c->sample1;
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nodes[0]->sample2 = c->sample2;
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if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
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version == AV_CODEC_ID_ADPCM_IMA_QT ||
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version == AV_CODEC_ID_ADPCM_SWF)
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nodes[0]->sample1 = c->prev_sample;
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if (version == AV_CODEC_ID_ADPCM_MS)
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nodes[0]->step = c->idelta;
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if (version == AV_CODEC_ID_ADPCM_YAMAHA) {
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if (c->step == 0) {
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nodes[0]->step = 127;
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nodes[0]->sample1 = 0;
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} else {
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nodes[0]->step = c->step;
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nodes[0]->sample1 = c->predictor;
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}
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}
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for (i = 0; i < n; i++) {
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TrellisNode *t = node_buf + frontier*(i&1);
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TrellisNode **u;
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int sample = samples[i * stride];
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int heap_pos = 0;
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memset(nodes_next, 0, frontier * sizeof(TrellisNode*));
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for (j = 0; j < frontier && nodes[j]; j++) {
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// higher j have higher ssd already, so they're likely
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// to yield a suboptimal next sample too
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const int range = (j < frontier / 2) ? 1 : 0;
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const int step = nodes[j]->step;
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int nidx;
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if (version == AV_CODEC_ID_ADPCM_MS) {
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const int predictor = ((nodes[j]->sample1 * c->coeff1) +
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(nodes[j]->sample2 * c->coeff2)) / 64;
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const int div = (sample - predictor) / step;
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const int nmin = av_clip(div-range, -8, 6);
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const int nmax = av_clip(div+range, -7, 7);
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for (nidx = nmin; nidx <= nmax; nidx++) {
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const int nibble = nidx & 0xf;
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int dec_sample = predictor + nidx * step;
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#define STORE_NODE(NAME, STEP_INDEX)\
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int d;\
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uint32_t ssd;\
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int pos;\
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TrellisNode *u;\
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uint8_t *h;\
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dec_sample = av_clip_int16(dec_sample);\
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d = sample - dec_sample;\
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ssd = nodes[j]->ssd + d*d;\
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/* Check for wraparound, skip such samples completely. \
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* Note, changing ssd to a 64 bit variable would be \
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* simpler, avoiding this check, but it's slower on \
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* x86 32 bit at the moment. */\
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if (ssd < nodes[j]->ssd)\
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goto next_##NAME;\
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/* Collapse any two states with the same previous sample value. \
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* One could also distinguish states by step and by 2nd to last
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* sample, but the effects of that are negligible.
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* Since nodes in the previous generation are iterated
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* through a heap, they're roughly ordered from better to
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* worse, but not strictly ordered. Therefore, an earlier
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* node with the same sample value is better in most cases
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* (and thus the current is skipped), but not strictly
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* in all cases. Only skipping samples where ssd >=
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* ssd of the earlier node with the same sample gives
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* slightly worse quality, though, for some reason. */ \
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h = &hash[(uint16_t) dec_sample];\
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if (*h == generation)\
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goto next_##NAME;\
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if (heap_pos < frontier) {\
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pos = heap_pos++;\
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} else {\
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/* Try to replace one of the leaf nodes with the new \
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* one, but try a different slot each time. */\
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pos = (frontier >> 1) +\
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(heap_pos & ((frontier >> 1) - 1));\
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if (ssd > nodes_next[pos]->ssd)\
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goto next_##NAME;\
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heap_pos++;\
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}\
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*h = generation;\
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u = nodes_next[pos];\
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if (!u) {\
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assert(pathn < FREEZE_INTERVAL << avctx->trellis);\
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u = t++;\
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nodes_next[pos] = u;\
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u->path = pathn++;\
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}\
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u->ssd = ssd;\
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u->step = STEP_INDEX;\
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u->sample2 = nodes[j]->sample1;\
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u->sample1 = dec_sample;\
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paths[u->path].nibble = nibble;\
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paths[u->path].prev = nodes[j]->path;\
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/* Sift the newly inserted node up in the heap to \
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* restore the heap property. */\
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while (pos > 0) {\
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int parent = (pos - 1) >> 1;\
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if (nodes_next[parent]->ssd <= ssd)\
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break;\
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FFSWAP(TrellisNode*, nodes_next[parent], nodes_next[pos]);\
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pos = parent;\
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}\
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next_##NAME:;
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STORE_NODE(ms, FFMAX(16,
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(ff_adpcm_AdaptationTable[nibble] * step) >> 8));
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}
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} else if (version == AV_CODEC_ID_ADPCM_IMA_WAV ||
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version == AV_CODEC_ID_ADPCM_IMA_QT ||
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version == AV_CODEC_ID_ADPCM_SWF) {
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#define LOOP_NODES(NAME, STEP_TABLE, STEP_INDEX)\
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const int predictor = nodes[j]->sample1;\
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const int div = (sample - predictor) * 4 / STEP_TABLE;\
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int nmin = av_clip(div - range, -7, 6);\
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int nmax = av_clip(div + range, -6, 7);\
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if (nmin <= 0)\
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nmin--; /* distinguish -0 from +0 */\
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if (nmax < 0)\
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nmax--;\
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for (nidx = nmin; nidx <= nmax; nidx++) {\
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const int nibble = nidx < 0 ? 7 - nidx : nidx;\
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int dec_sample = predictor +\
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(STEP_TABLE *\
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ff_adpcm_yamaha_difflookup[nibble]) / 8;\
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STORE_NODE(NAME, STEP_INDEX);\
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}
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LOOP_NODES(ima, ff_adpcm_step_table[step],
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av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
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} else { //AV_CODEC_ID_ADPCM_YAMAHA
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LOOP_NODES(yamaha, step,
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av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8,
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127, 24567));
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#undef LOOP_NODES
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#undef STORE_NODE
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}
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}
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u = nodes;
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nodes = nodes_next;
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nodes_next = u;
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generation++;
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if (generation == 255) {
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memset(hash, 0xff, 65536 * sizeof(*hash));
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generation = 0;
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}
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// prevent overflow
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if (nodes[0]->ssd > (1 << 28)) {
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for (j = 1; j < frontier && nodes[j]; j++)
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nodes[j]->ssd -= nodes[0]->ssd;
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nodes[0]->ssd = 0;
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}
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// merge old paths to save memory
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if (i == froze + FREEZE_INTERVAL) {
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p = &paths[nodes[0]->path];
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for (k = i; k > froze; k--) {
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dst[k] = p->nibble;
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p = &paths[p->prev];
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}
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froze = i;
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pathn = 0;
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// other nodes might use paths that don't coincide with the frozen one.
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// checking which nodes do so is too slow, so just kill them all.
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// this also slightly improves quality, but I don't know why.
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memset(nodes + 1, 0, (frontier - 1) * sizeof(TrellisNode*));
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}
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}
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p = &paths[nodes[0]->path];
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for (i = n - 1; i > froze; i--) {
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dst[i] = p->nibble;
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p = &paths[p->prev];
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}
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c->predictor = nodes[0]->sample1;
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c->sample1 = nodes[0]->sample1;
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c->sample2 = nodes[0]->sample2;
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c->step_index = nodes[0]->step;
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c->step = nodes[0]->step;
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c->idelta = nodes[0]->step;
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}
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|
|
|
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_packet(avpkt, pkt_size))) {
|
|
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
|
|
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][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 * 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_, \
|
|
.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");
|