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740 lines
28 KiB
740 lines
28 KiB
/* |
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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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|
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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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/** |
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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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* 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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|
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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 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 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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/* 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 + FF_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 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 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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#if FF_API_OLD_ENCODE_AUDIO |
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if (!(avctx->coded_frame = avcodec_alloc_frame())) |
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goto error; |
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#endif |
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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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#if FF_API_OLD_ENCODE_AUDIO |
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av_freep(&avctx->coded_frame); |
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#endif |
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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) |
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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 || |
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version == CODEC_ID_ADPCM_IMA_QT || |
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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 |
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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 == 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 == CODEC_ID_ADPCM_IMA_WAV || |
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version == CODEC_ID_ADPCM_IMA_QT || |
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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)\ |
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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 { //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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|
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u = nodes; |
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nodes = nodes_next; |
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nodes_next = u; |
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|
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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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|
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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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|
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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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|
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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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|
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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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|
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static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, |
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const AVFrame *frame, int *got_packet_ptr) |
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{ |
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int n, i, st, pkt_size, ret; |
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const int16_t *samples; |
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uint8_t *dst; |
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ADPCMEncodeContext *c = avctx->priv_data; |
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uint8_t *buf; |
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|
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samples = (const int16_t *)frame->data[0]; |
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st = avctx->channels == 2; |
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|
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if (avctx->codec_id == CODEC_ID_ADPCM_SWF) |
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pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8; |
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else |
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pkt_size = avctx->block_align; |
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if ((ret = ff_alloc_packet(avpkt, pkt_size))) { |
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); |
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return ret; |
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} |
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dst = avpkt->data; |
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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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n = frame->nb_samples / 8; |
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c->status[0].prev_sample = samples[0]; |
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/* c->status[0].step_index = 0; |
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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++ = 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 = 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++ = c->status[1].step_index; |
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*dst++ = 0; |
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samples++; |
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} |
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|
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/* stereo: 4 bytes (8 samples) for left, |
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4 bytes for right, 4 bytes left, ... */ |
|
if (avctx->trellis > 0) { |
|
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 8, error); |
|
adpcm_compress_trellis(avctx, samples, buf, &c->status[0], n * 8); |
|
if (avctx->channels == 2) |
|
adpcm_compress_trellis(avctx, samples + 1, buf + n * 8, |
|
&c->status[1], n * 8); |
|
for (i = 0; i < n; i++) { |
|
*dst++ = buf[8 * i + 0] | (buf[8 * i + 1] << 4); |
|
*dst++ = buf[8 * i + 2] | (buf[8 * i + 3] << 4); |
|
*dst++ = buf[8 * i + 4] | (buf[8 * i + 5] << 4); |
|
*dst++ = buf[8 * i + 6] | (buf[8 * i + 7] << 4); |
|
if (avctx->channels == 2) { |
|
uint8_t *buf1 = buf + n * 8; |
|
*dst++ = buf1[8 * i + 0] | (buf1[8 * i + 1] << 4); |
|
*dst++ = buf1[8 * i + 2] | (buf1[8 * i + 3] << 4); |
|
*dst++ = buf1[8 * i + 4] | (buf1[8 * i + 5] << 4); |
|
*dst++ = buf1[8 * i + 6] | (buf1[8 * i + 7] << 4); |
|
} |
|
} |
|
av_free(buf); |
|
} else { |
|
for (; n > 0; n--) { |
|
*dst = adpcm_ima_compress_sample(&c->status[0], samples[0]); |
|
*dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels ]) << 4; |
|
*dst = 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 = 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 = adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 6]); |
|
*dst++ |= adpcm_ima_compress_sample(&c->status[0], samples[avctx->channels * 7]) << 4; |
|
/* 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 = adpcm_ima_compress_sample(&c->status[1], samples[5 ]); |
|
*dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[7 ]) << 4; |
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[9 ]); |
|
*dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[11]) << 4; |
|
*dst = adpcm_ima_compress_sample(&c->status[1], samples[13]); |
|
*dst++ |= adpcm_ima_compress_sample(&c->status[1], samples[15]) << 4; |
|
} |
|
samples += 8 * avctx->channels; |
|
} |
|
} |
|
break; |
|
case CODEC_ID_ADPCM_IMA_QT: |
|
{ |
|
int ch, i; |
|
PutBitContext pb; |
|
init_put_bits(&pb, dst, pkt_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]); |
|
} else { |
|
for (i = 0; i < 64; i += 2) { |
|
int t1, t2; |
|
t1 = adpcm_ima_qt_compress_sample(&c->status[ch], |
|
samples[avctx->channels * (i + 0) + ch]); |
|
t2 = adpcm_ima_qt_compress_sample(&c->status[ch], |
|
samples[avctx->channels * (i + 1) + ch]); |
|
put_bits(&pb, 4, t2); |
|
put_bits(&pb, 4, t1); |
|
} |
|
} |
|
} |
|
|
|
flush_put_bits(&pb); |
|
break; |
|
} |
|
case CODEC_ID_ADPCM_SWF: |
|
{ |
|
int i; |
|
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 + 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 < 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 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 = 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); |
|
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: |
|
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 |
|
}; |
|
|
|
#define ADPCM_ENCODER(id_, name_, 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(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");
|
|
|