mirror of https://git.ffmpeg.org/ffmpeg.git
631 lines
19 KiB
C
631 lines
19 KiB
C
/*
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* Shorten decoder
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* Copyright (c) 2005 Jeff Muizelaar
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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* @file
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* Shorten decoder
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* @author Jeff Muizelaar
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*
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*/
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#include <limits.h>
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#include "avcodec.h"
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#include "bytestream.h"
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#include "get_bits.h"
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#include "golomb.h"
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#define MAX_CHANNELS 8
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#define MAX_BLOCKSIZE 65535
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#define OUT_BUFFER_SIZE 16384
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#define ULONGSIZE 2
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#define WAVE_FORMAT_PCM 0x0001
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#define DEFAULT_BLOCK_SIZE 256
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#define TYPESIZE 4
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#define CHANSIZE 0
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#define LPCQSIZE 2
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#define ENERGYSIZE 3
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#define BITSHIFTSIZE 2
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#define TYPE_S16HL 3
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#define TYPE_S16LH 5
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#define NWRAP 3
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#define NSKIPSIZE 1
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#define LPCQUANT 5
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#define V2LPCQOFFSET (1 << LPCQUANT)
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#define FNSIZE 2
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#define FN_DIFF0 0
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#define FN_DIFF1 1
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#define FN_DIFF2 2
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#define FN_DIFF3 3
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#define FN_QUIT 4
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#define FN_BLOCKSIZE 5
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#define FN_BITSHIFT 6
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#define FN_QLPC 7
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#define FN_ZERO 8
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#define FN_VERBATIM 9
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/** indicates if the FN_* command is audio or non-audio */
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static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 };
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#define VERBATIM_CKSIZE_SIZE 5
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#define VERBATIM_BYTE_SIZE 8
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#define CANONICAL_HEADER_SIZE 44
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typedef struct ShortenContext {
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AVCodecContext *avctx;
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GetBitContext gb;
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int min_framesize, max_framesize;
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int channels;
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int32_t *decoded[MAX_CHANNELS];
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int32_t *offset[MAX_CHANNELS];
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int *coeffs;
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uint8_t *bitstream;
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int bitstream_size;
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int bitstream_index;
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unsigned int allocated_bitstream_size;
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int header_size;
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uint8_t header[OUT_BUFFER_SIZE];
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int version;
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int cur_chan;
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int bitshift;
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int nmean;
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int internal_ftype;
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int nwrap;
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int blocksize;
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int bitindex;
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int32_t lpcqoffset;
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int got_header;
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int got_quit_command;
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} ShortenContext;
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static av_cold int shorten_decode_init(AVCodecContext * avctx)
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{
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ShortenContext *s = avctx->priv_data;
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s->avctx = avctx;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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return 0;
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}
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static int allocate_buffers(ShortenContext *s)
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{
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int i, chan;
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int *coeffs;
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void *tmp_ptr;
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for (chan=0; chan<s->channels; chan++) {
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if(FFMAX(1, s->nmean) >= UINT_MAX/sizeof(int32_t)){
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av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n");
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return -1;
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}
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if(s->blocksize + s->nwrap >= UINT_MAX/sizeof(int32_t) || s->blocksize + s->nwrap <= (unsigned)s->nwrap){
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av_log(s->avctx, AV_LOG_ERROR, "s->blocksize + s->nwrap too large\n");
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return -1;
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}
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tmp_ptr = av_realloc(s->offset[chan], sizeof(int32_t)*FFMAX(1, s->nmean));
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if (!tmp_ptr)
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return AVERROR(ENOMEM);
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s->offset[chan] = tmp_ptr;
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tmp_ptr = av_realloc(s->decoded[chan], sizeof(int32_t)*(s->blocksize + s->nwrap));
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if (!tmp_ptr)
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return AVERROR(ENOMEM);
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s->decoded[chan] = tmp_ptr;
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for (i=0; i<s->nwrap; i++)
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s->decoded[chan][i] = 0;
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s->decoded[chan] += s->nwrap;
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}
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coeffs = av_realloc(s->coeffs, s->nwrap * sizeof(*s->coeffs));
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if (!coeffs)
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return AVERROR(ENOMEM);
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s->coeffs = coeffs;
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return 0;
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}
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static inline unsigned int get_uint(ShortenContext *s, int k)
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{
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if (s->version != 0)
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k = get_ur_golomb_shorten(&s->gb, ULONGSIZE);
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return get_ur_golomb_shorten(&s->gb, k);
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}
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static void fix_bitshift(ShortenContext *s, int32_t *buffer)
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{
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int i;
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if (s->bitshift != 0)
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for (i = 0; i < s->blocksize; i++)
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buffer[i] <<= s->bitshift;
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}
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static void init_offset(ShortenContext *s)
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{
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int32_t mean = 0;
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int chan, i;
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int nblock = FFMAX(1, s->nmean);
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/* initialise offset */
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switch (s->internal_ftype)
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{
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case TYPE_S16HL:
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case TYPE_S16LH:
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mean = 0;
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break;
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default:
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av_log(s->avctx, AV_LOG_ERROR, "unknown audio type");
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abort();
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}
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for (chan = 0; chan < s->channels; chan++)
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for (i = 0; i < nblock; i++)
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s->offset[chan][i] = mean;
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}
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static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header,
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int header_size)
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{
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int len;
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short wave_format;
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if (bytestream_get_le32(&header) != MKTAG('R','I','F','F')) {
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av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
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return -1;
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}
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header += 4; /* chunk size */;
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if (bytestream_get_le32(&header) != MKTAG('W','A','V','E')) {
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av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n");
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return -1;
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}
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while (bytestream_get_le32(&header) != MKTAG('f','m','t',' ')) {
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len = bytestream_get_le32(&header);
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header += len;
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}
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len = bytestream_get_le32(&header);
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if (len < 16) {
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av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n");
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return -1;
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}
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wave_format = bytestream_get_le16(&header);
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switch (wave_format) {
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case WAVE_FORMAT_PCM:
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break;
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default:
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av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n");
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return -1;
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}
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header += 2; // skip channels (already got from shorten header)
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avctx->sample_rate = bytestream_get_le32(&header);
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header += 4; // skip bit rate (represents original uncompressed bit rate)
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header += 2; // skip block align (not needed)
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avctx->bits_per_coded_sample = bytestream_get_le16(&header);
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if (avctx->bits_per_coded_sample != 16) {
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av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample\n");
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return -1;
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}
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len -= 16;
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if (len > 0)
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av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len);
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return 0;
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}
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static void interleave_buffer(int16_t *samples, int nchan, int blocksize,
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int32_t **buffer)
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{
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int i, chan;
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for (i=0; i<blocksize; i++)
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for (chan=0; chan < nchan; chan++)
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*samples++ = av_clip_int16(buffer[chan][i]);
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}
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static const int fixed_coeffs[3][3] = {
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{ 1, 0, 0 },
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{ 2, -1, 0 },
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{ 3, -3, 1 }
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};
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static int decode_subframe_lpc(ShortenContext *s, int command, int channel,
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int residual_size, int32_t coffset)
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{
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int pred_order, sum, qshift, init_sum, i, j;
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const int *coeffs;
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if (command == FN_QLPC) {
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/* read/validate prediction order */
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pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE);
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if (pred_order > s->nwrap) {
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av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", pred_order);
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return AVERROR(EINVAL);
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}
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/* read LPC coefficients */
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for (i=0; i<pred_order; i++)
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s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT);
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coeffs = s->coeffs;
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qshift = LPCQUANT;
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} else {
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/* fixed LPC coeffs */
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pred_order = command;
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coeffs = fixed_coeffs[pred_order-1];
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qshift = 0;
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}
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/* subtract offset from previous samples to use in prediction */
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if (command == FN_QLPC && coffset)
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for (i = -pred_order; i < 0; i++)
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s->decoded[channel][i] -= coffset;
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/* decode residual and do LPC prediction */
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init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset;
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for (i=0; i < s->blocksize; i++) {
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sum = init_sum;
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for (j=0; j<pred_order; j++)
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sum += coeffs[j] * s->decoded[channel][i-j-1];
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s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) + (sum >> qshift);
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}
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/* add offset to current samples */
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if (command == FN_QLPC && coffset)
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for (i = 0; i < s->blocksize; i++)
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s->decoded[channel][i] += coffset;
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return 0;
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}
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static int read_header(ShortenContext *s)
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{
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int i, ret;
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int maxnlpc = 0;
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/* shorten signature */
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if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) {
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av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n");
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return -1;
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}
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s->lpcqoffset = 0;
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s->blocksize = DEFAULT_BLOCK_SIZE;
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s->channels = 1;
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s->nmean = -1;
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s->version = get_bits(&s->gb, 8);
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s->internal_ftype = get_uint(s, TYPESIZE);
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s->channels = get_uint(s, CHANSIZE);
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if (s->channels > MAX_CHANNELS) {
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av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
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return -1;
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}
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s->avctx->channels = s->channels;
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/* get blocksize if version > 0 */
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if (s->version > 0) {
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int skip_bytes, blocksize;
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blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE));
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if (!blocksize || blocksize > MAX_BLOCKSIZE) {
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av_log(s->avctx, AV_LOG_ERROR, "invalid or unsupported block size: %d\n",
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blocksize);
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return AVERROR(EINVAL);
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}
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s->blocksize = blocksize;
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maxnlpc = get_uint(s, LPCQSIZE);
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s->nmean = get_uint(s, 0);
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skip_bytes = get_uint(s, NSKIPSIZE);
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for (i=0; i<skip_bytes; i++) {
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skip_bits(&s->gb, 8);
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}
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}
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s->nwrap = FFMAX(NWRAP, maxnlpc);
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if ((ret = allocate_buffers(s)) < 0)
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return ret;
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init_offset(s);
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if (s->version > 1)
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s->lpcqoffset = V2LPCQOFFSET;
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if (get_ur_golomb_shorten(&s->gb, FNSIZE) != FN_VERBATIM) {
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av_log(s->avctx, AV_LOG_ERROR, "missing verbatim section at beginning of stream\n");
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return -1;
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}
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s->header_size = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
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if (s->header_size >= OUT_BUFFER_SIZE || s->header_size < CANONICAL_HEADER_SIZE) {
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av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n", s->header_size);
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return -1;
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}
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for (i=0; i<s->header_size; i++)
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s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
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if (decode_wave_header(s->avctx, s->header, s->header_size) < 0)
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return -1;
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s->cur_chan = 0;
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s->bitshift = 0;
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s->got_header = 1;
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return 0;
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}
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static int shorten_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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ShortenContext *s = avctx->priv_data;
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int i, input_buf_size = 0;
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int16_t *samples = data;
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int ret;
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/* allocate internal bitstream buffer */
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if(s->max_framesize == 0){
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void *tmp_ptr;
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s->max_framesize= 1024; // should hopefully be enough for the first header
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tmp_ptr = av_fast_realloc(s->bitstream, &s->allocated_bitstream_size,
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s->max_framesize);
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if (!tmp_ptr) {
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av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n");
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return AVERROR(ENOMEM);
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}
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s->bitstream = tmp_ptr;
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}
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/* append current packet data to bitstream buffer */
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if(1 && s->max_framesize){//FIXME truncated
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buf_size= FFMIN(buf_size, s->max_framesize - s->bitstream_size);
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input_buf_size= buf_size;
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if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
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memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
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s->bitstream_index=0;
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}
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if (buf)
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memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
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buf= &s->bitstream[s->bitstream_index];
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buf_size += s->bitstream_size;
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s->bitstream_size= buf_size;
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/* do not decode until buffer has at least max_framesize bytes or
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the end of the file has been reached */
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if (buf_size < s->max_framesize && avpkt->data) {
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*data_size = 0;
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return input_buf_size;
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}
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}
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/* init and position bitstream reader */
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init_get_bits(&s->gb, buf, buf_size*8);
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skip_bits(&s->gb, s->bitindex);
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/* process header or next subblock */
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if (!s->got_header) {
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if ((ret = read_header(s)) < 0)
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return ret;
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*data_size = 0;
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goto finish_frame;
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}
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|
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/* if quit command was read previously, don't decode anything */
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if (s->got_quit_command) {
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*data_size = 0;
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return avpkt->size;
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}
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s->cur_chan = 0;
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while (s->cur_chan < s->channels) {
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int cmd;
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int len;
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if (get_bits_left(&s->gb) < 3+FNSIZE) {
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*data_size = 0;
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break;
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}
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cmd = get_ur_golomb_shorten(&s->gb, FNSIZE);
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if (cmd > FN_VERBATIM) {
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av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd);
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*data_size = 0;
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break;
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}
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if (!is_audio_command[cmd]) {
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/* process non-audio command */
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switch (cmd) {
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case FN_VERBATIM:
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len = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
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while (len--) {
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get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
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}
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break;
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case FN_BITSHIFT:
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s->bitshift = get_ur_golomb_shorten(&s->gb, BITSHIFTSIZE);
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break;
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case FN_BLOCKSIZE: {
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int blocksize = get_uint(s, av_log2(s->blocksize));
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if (blocksize > s->blocksize) {
|
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av_log(avctx, AV_LOG_ERROR, "Increasing block size is not supported\n");
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return AVERROR_PATCHWELCOME;
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}
|
|
if (!blocksize || blocksize > MAX_BLOCKSIZE) {
|
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av_log(avctx, AV_LOG_ERROR, "invalid or unsupported "
|
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"block size: %d\n", blocksize);
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return AVERROR(EINVAL);
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}
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s->blocksize = blocksize;
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break;
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}
|
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case FN_QUIT:
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s->got_quit_command = 1;
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break;
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}
|
|
if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) {
|
|
*data_size = 0;
|
|
break;
|
|
}
|
|
} else {
|
|
/* process audio command */
|
|
int residual_size = 0;
|
|
int channel = s->cur_chan;
|
|
int32_t coffset;
|
|
|
|
/* get Rice code for residual decoding */
|
|
if (cmd != FN_ZERO) {
|
|
residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
|
|
/* this is a hack as version 0 differed in defintion of get_sr_golomb_shorten */
|
|
if (s->version == 0)
|
|
residual_size--;
|
|
}
|
|
|
|
/* calculate sample offset using means from previous blocks */
|
|
if (s->nmean == 0)
|
|
coffset = s->offset[channel][0];
|
|
else {
|
|
int32_t sum = (s->version < 2) ? 0 : s->nmean / 2;
|
|
for (i=0; i<s->nmean; i++)
|
|
sum += s->offset[channel][i];
|
|
coffset = sum / s->nmean;
|
|
if (s->version >= 2)
|
|
coffset >>= FFMIN(1, s->bitshift);
|
|
}
|
|
|
|
/* decode samples for this channel */
|
|
if (cmd == FN_ZERO) {
|
|
for (i=0; i<s->blocksize; i++)
|
|
s->decoded[channel][i] = 0;
|
|
} else {
|
|
if ((ret = decode_subframe_lpc(s, cmd, channel, residual_size, coffset)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
/* update means with info from the current block */
|
|
if (s->nmean > 0) {
|
|
int32_t sum = (s->version < 2) ? 0 : s->blocksize / 2;
|
|
for (i=0; i<s->blocksize; i++)
|
|
sum += s->decoded[channel][i];
|
|
|
|
for (i=1; i<s->nmean; i++)
|
|
s->offset[channel][i-1] = s->offset[channel][i];
|
|
|
|
if (s->version < 2)
|
|
s->offset[channel][s->nmean - 1] = sum / s->blocksize;
|
|
else
|
|
s->offset[channel][s->nmean - 1] = (sum / s->blocksize) << s->bitshift;
|
|
}
|
|
|
|
/* copy wrap samples for use with next block */
|
|
for (i=-s->nwrap; i<0; i++)
|
|
s->decoded[channel][i] = s->decoded[channel][i + s->blocksize];
|
|
|
|
/* shift samples to add in unused zero bits which were removed
|
|
during encoding */
|
|
fix_bitshift(s, s->decoded[channel]);
|
|
|
|
/* if this is the last channel in the block, output the samples */
|
|
s->cur_chan++;
|
|
if (s->cur_chan == s->channels) {
|
|
int out_size = s->blocksize * s->channels *
|
|
av_get_bytes_per_sample(avctx->sample_fmt);
|
|
if (*data_size < out_size) {
|
|
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
interleave_buffer(samples, s->channels, s->blocksize, s->decoded);
|
|
*data_size = out_size;
|
|
}
|
|
}
|
|
}
|
|
if (s->cur_chan < s->channels)
|
|
*data_size = 0;
|
|
|
|
finish_frame:
|
|
s->bitindex = get_bits_count(&s->gb) - 8*((get_bits_count(&s->gb))/8);
|
|
i= (get_bits_count(&s->gb))/8;
|
|
if (i > buf_size) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
|
|
s->bitstream_size=0;
|
|
s->bitstream_index=0;
|
|
return -1;
|
|
}
|
|
if (s->bitstream_size) {
|
|
s->bitstream_index += i;
|
|
s->bitstream_size -= i;
|
|
return input_buf_size;
|
|
} else
|
|
return i;
|
|
}
|
|
|
|
static av_cold int shorten_decode_close(AVCodecContext *avctx)
|
|
{
|
|
ShortenContext *s = avctx->priv_data;
|
|
int i;
|
|
|
|
for (i = 0; i < s->channels; i++) {
|
|
s->decoded[i] -= s->nwrap;
|
|
av_freep(&s->decoded[i]);
|
|
av_freep(&s->offset[i]);
|
|
}
|
|
av_freep(&s->bitstream);
|
|
av_freep(&s->coeffs);
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_shorten_decoder = {
|
|
.name = "shorten",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_SHORTEN,
|
|
.priv_data_size = sizeof(ShortenContext),
|
|
.init = shorten_decode_init,
|
|
.close = shorten_decode_close,
|
|
.decode = shorten_decode_frame,
|
|
.capabilities = CODEC_CAP_DELAY,
|
|
.long_name= NULL_IF_CONFIG_SMALL("Shorten"),
|
|
};
|