656 lines
26 KiB
C
656 lines
26 KiB
C
/*
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* Copyright (c) 2001-2003 The ffmpeg Project
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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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/**
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* @file
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* ADPCM encoders
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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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* Reference documents:
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* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
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* http://www.geocities.com/SiliconValley/8682/aud3.txt
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* http://openquicktime.sourceforge.net/plugins.htm
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* XAnim sources (xa_codec.c) http://www.rasnaimaging.com/people/lapus/download.html
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* http://www.cs.ucla.edu/~leec/mediabench/applications.html
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* SoX source code http://home.sprynet.com/~cbagwell/sox.html
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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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if (avctx->channels > 2)
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return -1; /* only stereo or mono =) */
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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 -1;
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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, max_paths * sizeof(*s->paths), error);
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FF_ALLOC_OR_GOTO(avctx, s->node_buf, 2 * frontier * sizeof(*s->node_buf), error);
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FF_ALLOC_OR_GOTO(avctx, s->nodep_buf, 2 * frontier * sizeof(*s->nodep_buf), error);
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FF_ALLOC_OR_GOTO(avctx, s->trellis_hash, 65536 * sizeof(*s->trellis_hash), error);
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}
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switch(avctx->codec->id) {
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case CODEC_ID_ADPCM_IMA_WAV:
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avctx->frame_size = (BLKSIZE - 4 * avctx->channels) * 8 / (4 * avctx->channels) + 1; /* each 16 bits sample gives one nibble */
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/* and we have 4 bytes per channel overhead */
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avctx->block_align = BLKSIZE;
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/* seems frame_size isn't taken into account... have to buffer the samples :-( */
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break;
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case 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 CODEC_ID_ADPCM_MS:
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avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2; /* each 16 bits sample gives one nibble */
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/* and we have 7 bytes per channel overhead */
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avctx->block_align = BLKSIZE;
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avctx->extradata_size = 32;
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extradata = avctx->extradata = av_malloc(avctx->extradata_size);
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if (!extradata)
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return AVERROR(ENOMEM);
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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 CODEC_ID_ADPCM_YAMAHA:
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avctx->frame_size = BLKSIZE * avctx->channels;
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avctx->block_align = BLKSIZE;
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break;
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case 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, 22050 or 44100\n");
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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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goto error;
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}
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avctx->coded_frame= avcodec_alloc_frame();
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avctx->coded_frame->key_frame= 1;
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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 -1;
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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(&avctx->coded_frame);
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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 unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, short 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/ff_adpcm_step_table[c->step_index]) + (delta<0)*8;
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c->prev_sample += ((ff_adpcm_step_table[c->step_index] * 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 unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, short sample)
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{
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int predictor, nibble, bias;
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predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
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nibble= sample - predictor;
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if(nibble>=0) bias= c->idelta/2;
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else 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 += (signed)((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[(int)nibble] * c->idelta) >> 8;
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if (c->idelta < 16) c->idelta = 16;
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return nibble;
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}
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static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c, short 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, const short *samples,
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uint8_t *dst, ADPCMChannelStatus *c, int n)
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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 stride = avctx->channels;
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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 == CODEC_ID_ADPCM_IMA_WAV) || (version == CODEC_ID_ADPCM_IMA_QT) || (version == CODEC_ID_ADPCM_SWF))
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nodes[0]->sample1 = c->prev_sample;
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if(version == CODEC_ID_ADPCM_MS)
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nodes[0]->step = c->idelta;
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if(version == 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 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 == CODEC_ID_ADPCM_MS) {
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const int predictor = ((nodes[j]->sample1 * c->coeff1) + (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) + (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, (ff_adpcm_AdaptationTable[nibble] * step) >> 8));
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}
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} else if((version == CODEC_ID_ADPCM_IMA_WAV)|| (version == CODEC_ID_ADPCM_IMA_QT)|| (version == 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) nmin--; /* distinguish -0 from +0 */\
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if(nmax<0) 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 + (STEP_TABLE * 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], av_clip(step + ff_adpcm_index_table[nibble], 0, 88));
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} else { //CODEC_ID_ADPCM_YAMAHA
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LOOP_NODES(yamaha, step, av_clip((step * ff_adpcm_yamaha_indexscale[nibble]) >> 8, 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,
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unsigned char *frame, int buf_size, void *data)
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{
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int n, i, st;
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short *samples;
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unsigned char *dst;
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ADPCMEncodeContext *c = avctx->priv_data;
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uint8_t *buf;
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dst = frame;
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samples = (short *)data;
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st= avctx->channels == 2;
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/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
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switch(avctx->codec->id) {
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case CODEC_ID_ADPCM_IMA_WAV:
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n = avctx->frame_size / 8;
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c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
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/* c->status[0].step_index = 0; *//* XXX: not sure how to init the state machine */
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bytestream_put_le16(&dst, c->status[0].prev_sample);
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*dst++ = (unsigned char)c->status[0].step_index;
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*dst++ = 0; /* unknown */
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samples++;
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if (avctx->channels == 2) {
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c->status[1].prev_sample = (signed short)samples[0];
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/* c->status[1].step_index = 0; */
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bytestream_put_le16(&dst, c->status[1].prev_sample);
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*dst++ = (unsigned char)c->status[1].step_index;
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*dst++ = 0;
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samples++;
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}
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/* stereo: 4 bytes (8 samples) for left, 4 bytes for right, 4 bytes left, ... */
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if(avctx->trellis > 0) {
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FF_ALLOC_OR_GOTO(avctx, buf, 2*n*8, error);
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adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n*8);
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if(avctx->channels == 2)
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adpcm_compress_trellis(avctx, samples+1, buf + n*8, &c->status[1], n*8);
|
|
for(i=0; i<n; i++) {
|
|
*dst++ = buf[8*i+0] | (buf[8*i+1] << 4);
|
|
*dst++ = buf[8*i+2] | (buf[8*i+3] << 4);
|
|
*dst++ = buf[8*i+4] | (buf[8*i+5] << 4);
|
|
*dst++ = buf[8*i+6] | (buf[8*i+7] << 4);
|
|
if (avctx->channels == 2) {
|
|
uint8_t *buf1 = buf + n*8;
|
|
*dst++ = buf1[8*i+0] | (buf1[8*i+1] << 4);
|
|
*dst++ = buf1[8*i+2] | (buf1[8*i+3] << 4);
|
|
*dst++ = buf1[8*i+4] | (buf1[8*i+5] << 4);
|
|
*dst++ = buf1[8*i+6] | (buf1[8*i+7] << 4);
|
|
}
|
|
}
|
|
av_free(buf);
|
|
} else
|
|
for (; n>0; n--) {
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[0]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 2]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 3]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 4]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 5]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4;
|
|
dst++;
|
|
/* right channel */
|
|
if (avctx->channels == 2) {
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[1]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[3]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[5]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[7]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[9]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4;
|
|
dst++;
|
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[13]);
|
|
*dst |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4;
|
|
dst++;
|
|
}
|
|
samples += 8 * avctx->channels;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_IMA_QT:
|
|
{
|
|
int ch, i;
|
|
PutBitContext pb;
|
|
init_put_bits(&pb, dst, buf_size*8);
|
|
|
|
for(ch=0; ch<avctx->channels; ch++){
|
|
put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
|
|
put_bits(&pb, 7, c->status[ch].step_index);
|
|
if(avctx->trellis > 0) {
|
|
uint8_t buf[64];
|
|
adpcm_compress_trellis(avctx, samples+ch, buf, &c->status[ch], 64);
|
|
for(i=0; i<64; i++)
|
|
put_bits(&pb, 4, buf[i^1]);
|
|
c->status[ch].prev_sample = c->status[ch].predictor & ~0x7F;
|
|
} else {
|
|
for (i=0; i<64; i+=2){
|
|
int t1, t2;
|
|
t1 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+0)+ch]);
|
|
t2 = adpcm_ima_compress_sample(&c->status[ch], samples[avctx->channels*(i+1)+ch]);
|
|
put_bits(&pb, 4, t2);
|
|
put_bits(&pb, 4, t1);
|
|
}
|
|
c->status[ch].prev_sample &= ~0x7F;
|
|
}
|
|
}
|
|
|
|
flush_put_bits(&pb);
|
|
dst += put_bits_count(&pb)>>3;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_SWF:
|
|
{
|
|
int i;
|
|
PutBitContext pb;
|
|
init_put_bits(&pb, dst, buf_size*8);
|
|
|
|
n = avctx->frame_size-1;
|
|
|
|
//Store AdpcmCodeSize
|
|
put_bits(&pb, 2, 2); //Set 4bits flash adpcm format
|
|
|
|
//Init the encoder state
|
|
for(i=0; i<avctx->channels; i++){
|
|
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63); // clip step so it fits 6 bits
|
|
put_sbits(&pb, 16, samples[i]);
|
|
put_bits(&pb, 6, c->status[i].step_index);
|
|
c->status[i].prev_sample = (signed short)samples[i];
|
|
}
|
|
|
|
if(avctx->trellis > 0) {
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
|
|
adpcm_compress_trellis(avctx, samples+2, buf, &c->status[0], n);
|
|
if (avctx->channels == 2)
|
|
adpcm_compress_trellis(avctx, samples+3, buf+n, &c->status[1], n);
|
|
for(i=0; i<n; i++) {
|
|
put_bits(&pb, 4, buf[i]);
|
|
if (avctx->channels == 2)
|
|
put_bits(&pb, 4, buf[n+i]);
|
|
}
|
|
av_free(buf);
|
|
} else {
|
|
for (i=1; i<avctx->frame_size; i++) {
|
|
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels*i]));
|
|
if (avctx->channels == 2)
|
|
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[1], samples[2*i+1]));
|
|
}
|
|
}
|
|
flush_put_bits(&pb);
|
|
dst += put_bits_count(&pb)>>3;
|
|
break;
|
|
}
|
|
case CODEC_ID_ADPCM_MS:
|
|
for(i=0; i<avctx->channels; i++){
|
|
int predictor=0;
|
|
|
|
*dst++ = predictor;
|
|
c->status[i].coeff1 = ff_adpcm_AdaptCoeff1[predictor];
|
|
c->status[i].coeff2 = ff_adpcm_AdaptCoeff2[predictor];
|
|
}
|
|
for(i=0; i<avctx->channels; i++){
|
|
if (c->status[i].idelta < 16)
|
|
c->status[i].idelta = 16;
|
|
|
|
bytestream_put_le16(&dst, c->status[i].idelta);
|
|
}
|
|
for(i=0; i<avctx->channels; i++){
|
|
c->status[i].sample2= *samples++;
|
|
}
|
|
for(i=0; i<avctx->channels; i++){
|
|
c->status[i].sample1= *samples++;
|
|
|
|
bytestream_put_le16(&dst, c->status[i].sample1);
|
|
}
|
|
for(i=0; i<avctx->channels; i++)
|
|
bytestream_put_le16(&dst, c->status[i].sample2);
|
|
|
|
if(avctx->trellis > 0) {
|
|
int n = avctx->block_align - 7*avctx->channels;
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n, error);
|
|
if(avctx->channels == 1) {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
for(i=0; i<n; i+=2)
|
|
*dst++ = (buf[i] << 4) | buf[i+1];
|
|
} else {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
|
|
for(i=0; i<n; i++)
|
|
*dst++ = (buf[i] << 4) | buf[n+i];
|
|
}
|
|
av_free(buf);
|
|
} else
|
|
for(i=7*avctx->channels; i<avctx->block_align; i++) {
|
|
int nibble;
|
|
nibble = adpcm_ms_compress_sample(&c->status[ 0], *samples++)<<4;
|
|
nibble|= adpcm_ms_compress_sample(&c->status[st], *samples++);
|
|
*dst++ = nibble;
|
|
}
|
|
break;
|
|
case CODEC_ID_ADPCM_YAMAHA:
|
|
n = avctx->frame_size / 2;
|
|
if(avctx->trellis > 0) {
|
|
FF_ALLOC_OR_GOTO(avctx, buf, 2*n*2, error);
|
|
n *= 2;
|
|
if(avctx->channels == 1) {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
for(i=0; i<n; i+=2)
|
|
*dst++ = buf[i] | (buf[i+1] << 4);
|
|
} else {
|
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n);
|
|
adpcm_compress_trellis(avctx, samples+1, buf+n, &c->status[1], n);
|
|
for(i=0; i<n; i++)
|
|
*dst++ = buf[i] | (buf[n+i] << 4);
|
|
}
|
|
av_free(buf);
|
|
} else
|
|
for (n *= avctx->channels; n>0; n--) {
|
|
int nibble;
|
|
nibble = adpcm_yamaha_compress_sample(&c->status[ 0], *samples++);
|
|
nibble |= adpcm_yamaha_compress_sample(&c->status[st], *samples++) << 4;
|
|
*dst++ = nibble;
|
|
}
|
|
break;
|
|
default:
|
|
error:
|
|
return -1;
|
|
}
|
|
return dst - frame;
|
|
}
|
|
|
|
|
|
#define ADPCM_ENCODER(id,name,long_name_) \
|
|
AVCodec ff_ ## name ## _encoder = { \
|
|
#name, \
|
|
AVMEDIA_TYPE_AUDIO, \
|
|
id, \
|
|
sizeof(ADPCMEncodeContext), \
|
|
adpcm_encode_init, \
|
|
adpcm_encode_frame, \
|
|
adpcm_encode_close, \
|
|
NULL, \
|
|
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, \
|
|
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
|
|
}
|
|
|
|
ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_QT, adpcm_ima_qt, "ADPCM IMA QuickTime");
|
|
ADPCM_ENCODER(CODEC_ID_ADPCM_IMA_WAV, adpcm_ima_wav, "ADPCM IMA WAV");
|
|
ADPCM_ENCODER(CODEC_ID_ADPCM_MS, adpcm_ms, "ADPCM Microsoft");
|
|
ADPCM_ENCODER(CODEC_ID_ADPCM_SWF, adpcm_swf, "ADPCM Shockwave Flash");
|
|
ADPCM_ENCODER(CODEC_ID_ADPCM_YAMAHA, adpcm_yamaha, "ADPCM Yamaha");
|