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ffmpeg/libavcodec/shorten.c
Luca Barbato 5cf7c72757 shorten: use the unsigned type where needed 
get_uint returns an unsigned value, use an unsigned to store
blocksize to make sure the comparison logic is correct and report
correctly the error for the channel count not supported.
2013-03-06 01:04:02 +01:00

657 lines
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/*
* Shorten decoder
* Copyright (c) 2005 Jeff Muizelaar
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Shorten decoder
* @author Jeff Muizelaar
*
*/
#include <limits.h>
#include "avcodec.h"
#include "bytestream.h"
#include "get_bits.h"
#include "golomb.h"
#include "internal.h"
#define MAX_CHANNELS 8
#define MAX_BLOCKSIZE 65535
#define OUT_BUFFER_SIZE 16384
#define ULONGSIZE 2
#define WAVE_FORMAT_PCM 0x0001
#define DEFAULT_BLOCK_SIZE 256
#define TYPESIZE 4
#define CHANSIZE 0
#define LPCQSIZE 2
#define ENERGYSIZE 3
#define BITSHIFTSIZE 2
#define TYPE_S16HL 3
#define TYPE_S16LH 5
#define NWRAP 3
#define NSKIPSIZE 1
#define LPCQUANT 5
#define V2LPCQOFFSET (1 << LPCQUANT)
#define FNSIZE 2
#define FN_DIFF0 0
#define FN_DIFF1 1
#define FN_DIFF2 2
#define FN_DIFF3 3
#define FN_QUIT 4
#define FN_BLOCKSIZE 5
#define FN_BITSHIFT 6
#define FN_QLPC 7
#define FN_ZERO 8
#define FN_VERBATIM 9
/** indicates if the FN_* command is audio or non-audio */
static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 };
#define VERBATIM_CKSIZE_SIZE 5
#define VERBATIM_BYTE_SIZE 8
#define CANONICAL_HEADER_SIZE 44
typedef struct ShortenContext {
AVCodecContext *avctx;
GetBitContext gb;
int min_framesize, max_framesize;
unsigned channels;
int32_t *decoded[MAX_CHANNELS];
int32_t *decoded_base[MAX_CHANNELS];
int32_t *offset[MAX_CHANNELS];
int *coeffs;
uint8_t *bitstream;
int bitstream_size;
int bitstream_index;
unsigned int allocated_bitstream_size;
int header_size;
uint8_t header[OUT_BUFFER_SIZE];
int version;
int cur_chan;
int bitshift;
int nmean;
int internal_ftype;
int nwrap;
int blocksize;
int bitindex;
int32_t lpcqoffset;
int got_header;
int got_quit_command;
} ShortenContext;
static av_cold int shorten_decode_init(AVCodecContext *avctx)
{
ShortenContext *s = avctx->priv_data;
s->avctx = avctx;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
return 0;
}
static int allocate_buffers(ShortenContext *s)
{
int i, chan;
int *coeffs;
void *tmp_ptr;
for (chan = 0; chan < s->channels; chan++) {
if (FFMAX(1, s->nmean) >= UINT_MAX / sizeof(int32_t)) {
av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n");
return AVERROR_INVALIDDATA;
}
if (s->blocksize + s->nwrap >= UINT_MAX / sizeof(int32_t) ||
s->blocksize + s->nwrap <= (unsigned)s->nwrap) {
av_log(s->avctx, AV_LOG_ERROR,
"s->blocksize + s->nwrap too large\n");
return AVERROR_INVALIDDATA;
}
tmp_ptr =
av_realloc(s->offset[chan], sizeof(int32_t) * FFMAX(1, s->nmean));
if (!tmp_ptr)
return AVERROR(ENOMEM);
s->offset[chan] = tmp_ptr;
tmp_ptr = av_realloc(s->decoded_base[chan], (s->blocksize + s->nwrap) *
sizeof(s->decoded_base[0][0]));
if (!tmp_ptr)
return AVERROR(ENOMEM);
s->decoded_base[chan] = tmp_ptr;
for (i = 0; i < s->nwrap; i++)
s->decoded_base[chan][i] = 0;
s->decoded[chan] = s->decoded_base[chan] + s->nwrap;
}
coeffs = av_realloc(s->coeffs, s->nwrap * sizeof(*s->coeffs));
if (!coeffs)
return AVERROR(ENOMEM);
s->coeffs = coeffs;
return 0;
}
static inline unsigned int get_uint(ShortenContext *s, int k)
{
if (s->version != 0)
k = get_ur_golomb_shorten(&s->gb, ULONGSIZE);
return get_ur_golomb_shorten(&s->gb, k);
}
static void fix_bitshift(ShortenContext *s, int32_t *buffer)
{
int i;
if (s->bitshift != 0)
for (i = 0; i < s->blocksize; i++)
buffer[i] <<= s->bitshift;
}
static int init_offset(ShortenContext *s)
{
int32_t mean = 0;
int chan, i;
int nblock = FFMAX(1, s->nmean);
/* initialise offset */
switch (s->internal_ftype) {
case TYPE_S16HL:
case TYPE_S16LH:
mean = 0;
break;
default:
av_log(s->avctx, AV_LOG_ERROR, "unknown audio type");
return AVERROR_INVALIDDATA;
}
for (chan = 0; chan < s->channels; chan++)
for (i = 0; i < nblock; i++)
s->offset[chan][i] = mean;
return 0;
}
static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header,
int header_size)
{
int len;
short wave_format;
if (bytestream_get_le32(&header) != MKTAG('R', 'I', 'F', 'F')) {
av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
return AVERROR_INVALIDDATA;
}
header += 4; /* chunk size */
if (bytestream_get_le32(&header) != MKTAG('W', 'A', 'V', 'E')) {
av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n");
return AVERROR_INVALIDDATA;
}
while (bytestream_get_le32(&header) != MKTAG('f', 'm', 't', ' ')) {
len = bytestream_get_le32(&header);
header += len;
}
len = bytestream_get_le32(&header);
if (len < 16) {
av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n");
return AVERROR_INVALIDDATA;
}
wave_format = bytestream_get_le16(&header);
switch (wave_format) {
case WAVE_FORMAT_PCM:
break;
default:
av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n");
return AVERROR(ENOSYS);
}
header += 2; // skip channels (already got from shorten header)
avctx->sample_rate = bytestream_get_le32(&header);
header += 4; // skip bit rate (represents original uncompressed bit rate)
header += 2; // skip block align (not needed)
avctx->bits_per_coded_sample = bytestream_get_le16(&header);
if (avctx->bits_per_coded_sample != 16) {
av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample\n");
return AVERROR(ENOSYS);
}
len -= 16;
if (len > 0)
av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len);
return 0;
}
static void output_buffer(int16_t **samples, int nchan, int blocksize,
int32_t **buffer)
{
int i, ch;
for (ch = 0; ch < nchan; ch++) {
int32_t *in = buffer[ch];
int16_t *out = samples[ch];
for (i = 0; i < blocksize; i++)
out[i] = av_clip_int16(in[i]);
}
}
static const int fixed_coeffs[3][3] = {
{ 1, 0, 0 },
{ 2, -1, 0 },
{ 3, -3, 1 }
};
static int decode_subframe_lpc(ShortenContext *s, int command, int channel,
int residual_size, int32_t coffset)
{
int pred_order, sum, qshift, init_sum, i, j;
const int *coeffs;
if (command == FN_QLPC) {
/* read/validate prediction order */
pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE);
if (pred_order > s->nwrap) {
av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n",
pred_order);
return AVERROR(EINVAL);
}
/* read LPC coefficients */
for (i = 0; i < pred_order; i++)
s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT);
coeffs = s->coeffs;
qshift = LPCQUANT;
} else {
/* fixed LPC coeffs */
pred_order = command;
coeffs = fixed_coeffs[pred_order - 1];
qshift = 0;
}
/* subtract offset from previous samples to use in prediction */
if (command == FN_QLPC && coffset)
for (i = -pred_order; i < 0; i++)
s->decoded[channel][i] -= coffset;
/* decode residual and do LPC prediction */
init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset;
for (i = 0; i < s->blocksize; i++) {
sum = init_sum;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * s->decoded[channel][i - j - 1];
s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) +
(sum >> qshift);
}
/* add offset to current samples */
if (command == FN_QLPC && coffset)
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] += coffset;
return 0;
}
static int read_header(ShortenContext *s)
{
int i, ret;
int maxnlpc = 0;
/* shorten signature */
if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) {
av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n");
return AVERROR_INVALIDDATA;
}
s->lpcqoffset = 0;
s->blocksize = DEFAULT_BLOCK_SIZE;
s->nmean = -1;
s->version = get_bits(&s->gb, 8);
s->internal_ftype = get_uint(s, TYPESIZE);
s->channels = get_uint(s, CHANSIZE);
if (!s->channels) {
av_log(s->avctx, AV_LOG_ERROR, "No channels reported\n");
return AVERROR_INVALIDDATA;
}
if (s->channels > MAX_CHANNELS) {
av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
s->channels = 0;
return AVERROR_INVALIDDATA;
}
s->avctx->channels = s->channels;
/* get blocksize if version > 0 */
if (s->version > 0) {
int skip_bytes;
unsigned blocksize;
blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE));
if (!blocksize || blocksize > MAX_BLOCKSIZE) {
av_log(s->avctx, AV_LOG_ERROR,
"invalid or unsupported block size: %d\n",
blocksize);
return AVERROR(EINVAL);
}
s->blocksize = blocksize;
maxnlpc = get_uint(s, LPCQSIZE);
s->nmean = get_uint(s, 0);
skip_bytes = get_uint(s, NSKIPSIZE);
for (i = 0; i < skip_bytes; i++)
skip_bits(&s->gb, 8);
}
s->nwrap = FFMAX(NWRAP, maxnlpc);
if ((ret = allocate_buffers(s)) < 0)
return ret;
if ((ret = init_offset(s)) < 0)
return ret;
if (s->version > 1)
s->lpcqoffset = V2LPCQOFFSET;
if (get_ur_golomb_shorten(&s->gb, FNSIZE) != FN_VERBATIM) {
av_log(s->avctx, AV_LOG_ERROR,
"missing verbatim section at beginning of stream\n");
return AVERROR_INVALIDDATA;
}
s->header_size = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
if (s->header_size >= OUT_BUFFER_SIZE ||
s->header_size < CANONICAL_HEADER_SIZE) {
av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n",
s->header_size);
return AVERROR_INVALIDDATA;
}
for (i = 0; i < s->header_size; i++)
s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
if ((ret = decode_wave_header(s->avctx, s->header, s->header_size)) < 0)
return ret;
s->cur_chan = 0;
s->bitshift = 0;
s->got_header = 1;
return 0;
}
static int shorten_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ShortenContext *s = avctx->priv_data;
int i, input_buf_size = 0;
int ret;
/* allocate internal bitstream buffer */
if (s->max_framesize == 0) {
void *tmp_ptr;
s->max_framesize = 1024; // should hopefully be enough for the first header
tmp_ptr = av_fast_realloc(s->bitstream, &s->allocated_bitstream_size,
s->max_framesize);
if (!tmp_ptr) {
av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n");
return AVERROR(ENOMEM);
}
s->bitstream = tmp_ptr;
}
/* append current packet data to bitstream buffer */
if (1 && s->max_framesize) { //FIXME truncated
buf_size = FFMIN(buf_size, s->max_framesize - s->bitstream_size);
input_buf_size = buf_size;
if (s->bitstream_index + s->bitstream_size + buf_size >
s->allocated_bitstream_size) {
memmove(s->bitstream, &s->bitstream[s->bitstream_index],
s->bitstream_size);
s->bitstream_index = 0;
}
if (buf)
memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf,
buf_size);
buf = &s->bitstream[s->bitstream_index];
buf_size += s->bitstream_size;
s->bitstream_size = buf_size;
/* do not decode until buffer has at least max_framesize bytes or
* the end of the file has been reached */
if (buf_size < s->max_framesize && avpkt->data) {
*got_frame_ptr = 0;
return input_buf_size;
}
}
/* init and position bitstream reader */
init_get_bits(&s->gb, buf, buf_size * 8);
skip_bits(&s->gb, s->bitindex);
/* process header or next subblock */
if (!s->got_header) {
if ((ret = read_header(s)) < 0)
return ret;
*got_frame_ptr = 0;
goto finish_frame;
}
/* if quit command was read previously, don't decode anything */
if (s->got_quit_command) {
*got_frame_ptr = 0;
return avpkt->size;
}
s->cur_chan = 0;
while (s->cur_chan < s->channels) {
unsigned cmd;
int len;
if (get_bits_left(&s->gb) < 3 + FNSIZE) {
*got_frame_ptr = 0;
break;
}
cmd = get_ur_golomb_shorten(&s->gb, FNSIZE);
if (cmd > FN_VERBATIM) {
av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd);
*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 >>= 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) {
/* get output buffer */
frame->nb_samples = s->blocksize;
if ((ret = ff_get_buffer(avctx, frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
/* interleave output */
output_buffer((int16_t **)frame->extended_data, s->channels,
s->blocksize, s->decoded);
*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_NONE },
};
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