mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
4caf8c51de
* commit 'b26742cc308552f242ee2bf93b07a3ff509f4edc':
shorten: Break out of loop looking for fmt chunk if none is found
See: 5e9a56a035
Merged-by: Michael Niedermayer <michaelni@gmx.at>
676 lines
21 KiB
C
676 lines
21 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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#include "internal.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_S8 1
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#define TYPE_U8 2
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#define TYPE_S16HL 3
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#define TYPE_U16HL 4
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#define TYPE_S16LH 5
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#define TYPE_U16LH 6
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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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unsigned channels;
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int32_t *decoded[MAX_CHANNELS];
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int32_t *decoded_base[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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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 AVERROR_INVALIDDATA;
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}
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if (s->blocksize + s->nwrap >= UINT_MAX / sizeof(int32_t) ||
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s->blocksize + s->nwrap <= (unsigned)s->nwrap) {
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av_log(s->avctx, AV_LOG_ERROR,
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"s->blocksize + s->nwrap too large\n");
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return AVERROR_INVALIDDATA;
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}
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tmp_ptr =
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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_base[chan], (s->blocksize + s->nwrap) *
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sizeof(s->decoded_base[0][0]));
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if (!tmp_ptr)
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return AVERROR(ENOMEM);
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s->decoded_base[chan] = tmp_ptr;
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for (i = 0; i < s->nwrap; i++)
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s->decoded_base[chan][i] = 0;
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s->decoded[chan] = s->decoded_base[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 int 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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case TYPE_U8:
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s->avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
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mean = 0x80;
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break;
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case TYPE_S16HL:
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case TYPE_S16LH:
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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break;
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default:
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av_log(s->avctx, AV_LOG_ERROR, "unknown audio type\n");
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return AVERROR_PATCHWELCOME;
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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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return 0;
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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, bps;
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short wave_format;
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GetByteContext gb;
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bytestream2_init(&gb, header, header_size);
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if (bytestream2_get_le32(&gb) != MKTAG('R', 'I', 'F', 'F')) {
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av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
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return AVERROR_INVALIDDATA;
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}
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bytestream2_skip(&gb, 4); /* chunk size */
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if (bytestream2_get_le32(&gb) != MKTAG('W', 'A', 'V', 'E')) {
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av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n");
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return AVERROR_INVALIDDATA;
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}
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while (bytestream2_get_le32(&gb) != MKTAG('f', 'm', 't', ' ')) {
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len = bytestream2_get_le32(&gb);
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bytestream2_skip(&gb, len);
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if (len < 0 || bytestream2_get_bytes_left(&gb) < 16) {
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av_log(avctx, AV_LOG_ERROR, "no fmt chunk found\n");
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return AVERROR_INVALIDDATA;
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}
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}
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len = bytestream2_get_le32(&gb);
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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 AVERROR_INVALIDDATA;
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}
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wave_format = bytestream2_get_le16(&gb);
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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 AVERROR(ENOSYS);
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}
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bytestream2_skip(&gb, 2); // skip channels (already got from shorten header)
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avctx->sample_rate = bytestream2_get_le32(&gb);
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bytestream2_skip(&gb, 4); // skip bit rate (represents original uncompressed bit rate)
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bytestream2_skip(&gb, 2); // skip block align (not needed)
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bps = bytestream2_get_le16(&gb);
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avctx->bits_per_coded_sample = bps;
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if (bps != 16 && bps != 8) {
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av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample: %d\n", bps);
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return AVERROR(ENOSYS);
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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 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",
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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) +
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(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 AVERROR_INVALIDDATA;
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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->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) {
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av_log(s->avctx, AV_LOG_ERROR, "No channels reported\n");
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return AVERROR_INVALIDDATA;
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}
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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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s->channels = 0;
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return AVERROR_INVALIDDATA;
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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;
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unsigned 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,
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"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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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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if ((ret = init_offset(s)) < 0)
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return ret;
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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,
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"missing verbatim section at beginning of stream\n");
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return AVERROR_INVALIDDATA;
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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 ||
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s->header_size < CANONICAL_HEADER_SIZE) {
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av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n",
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s->header_size);
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return AVERROR_INVALIDDATA;
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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 ((ret = decode_wave_header(s->avctx, s->header, s->header_size)) < 0)
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return ret;
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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, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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AVFrame *frame = data;
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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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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 = 8192; // 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 + FF_INPUT_BUFFER_PADDING_SIZE);
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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 + FF_INPUT_BUFFER_PADDING_SIZE >
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s->allocated_bitstream_size) {
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memmove(s->bitstream, &s->bitstream[s->bitstream_index],
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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,
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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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*got_frame_ptr = 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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*got_frame_ptr = 0;
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goto finish_frame;
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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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*got_frame_ptr = 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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unsigned cmd;
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int len;
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if (get_bits_left(&s->gb) < 3 + FNSIZE) {
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*got_frame_ptr = 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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*got_frame_ptr = 0;
|
|
break;
|
|
}
|
|
|
|
if (!is_audio_command[cmd]) {
|
|
/* process non-audio command */
|
|
switch (cmd) {
|
|
case FN_VERBATIM:
|
|
len = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
|
|
while (len--)
|
|
get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
|
|
break;
|
|
case FN_BITSHIFT:
|
|
s->bitshift = get_ur_golomb_shorten(&s->gb, BITSHIFTSIZE);
|
|
break;
|
|
case FN_BLOCKSIZE: {
|
|
unsigned blocksize = get_uint(s, av_log2(s->blocksize));
|
|
if (blocksize > s->blocksize) {
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"Increasing block size is not supported\n");
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
if (!blocksize || blocksize > MAX_BLOCKSIZE) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid or unsupported "
|
|
"block size: %d\n", blocksize);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->blocksize = blocksize;
|
|
break;
|
|
}
|
|
case FN_QUIT:
|
|
s->got_quit_command = 1;
|
|
break;
|
|
}
|
|
if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) {
|
|
*got_frame_ptr = 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 the definition
|
|
* 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 = s->bitshift == 0 ? coffset : coffset >> s->bitshift - 1 >> 1;
|
|
}
|
|
|
|
/* 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) {
|
|
uint8_t *samples_u8;
|
|
int16_t *samples_s16;
|
|
int chan;
|
|
|
|
/* get output buffer */
|
|
frame->nb_samples = s->blocksize;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
|
|
for (chan = 0; chan < s->channels; chan++) {
|
|
samples_u8 = ((uint8_t **)frame->extended_data)[chan];
|
|
samples_s16 = ((int16_t **)frame->extended_data)[chan];
|
|
for (i = 0; i < s->blocksize; i++) {
|
|
switch (s->internal_ftype) {
|
|
case TYPE_U8:
|
|
*samples_u8++ = av_clip_uint8(s->decoded[chan][i]);
|
|
break;
|
|
case TYPE_S16HL:
|
|
case TYPE_S16LH:
|
|
*samples_s16++ = av_clip_int16(s->decoded[chan][i]);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
}
|
|
}
|
|
}
|
|
if (s->cur_chan < s->channels)
|
|
*got_frame_ptr = 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 AVERROR_INVALIDDATA;
|
|
}
|
|
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] = NULL;
|
|
av_freep(&s->decoded_base[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 = AV_CODEC_ID_SHORTEN,
|
|
.priv_data_size = sizeof(ShortenContext),
|
|
.init = shorten_decode_init,
|
|
.close = shorten_decode_close,
|
|
.decode = shorten_decode_frame,
|
|
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
|
|
.long_name = NULL_IF_CONFIG_SMALL("Shorten"),
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
|
|
AV_SAMPLE_FMT_U8P,
|
|
AV_SAMPLE_FMT_NONE },
|
|
};
|