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ffmpeg/libavcodec/shorten.c
Michael Niedermayer 8c1ebdcea2 Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  shorten: Use separate pointers for the allocated memory for decoded samples.
  atrac3: Fix crash in tonal component decoding.
  ws_snd1: Fix wrong samples counts.
  movenc: Don't set a default sample duration when creating ismv
  rtp: Factorize the check for distinguishing RTCP packets from RTP
  golomb: avoid infinite loop on all-zero input (or end of buffer).
  bethsoftvid: synchronize video timestamps with audio sample rate
  bethsoftvid: add audio stream only after getting the first audio packet
  bethsoftvid: Set video packet duration instead of accumulating pts.
  bethsoftvid: set packet key frame flag for audio and I-frame video packets.
  bethsoftvid: fix read_packet() return codes.
  bethsoftvid: pass palette in side data instead of in a separate packet.
  sdp: Ignore RTCP packets when autodetecting RTP streams
  proresenc: initialise 'sign' variable
  mpegaudio: replace memcpy by SIMD code
  vc1: prevent using last_frame as a reference for I/P first frame.

Conflicts:
	libavcodec/atrac3.c
	libavcodec/golomb.h
	libavcodec/shorten.c
	libavcodec/ws-snd1.c
	tests/ref/fate/bethsoft-vid

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-02-17 00:35:06 +01:00

643 lines
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/*
* Shorten decoder
* Copyright (c) 2005 Jeff Muizelaar
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; 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"
#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;
AVFrame frame;
GetBitContext gb;
int min_framesize, max_framesize;
int 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_S16;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
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 -1;
}
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 -1;
}
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;
const uint8_t *end= header + header_size;
if (bytestream_get_le32(&header) != MKTAG('R','I','F','F')) {
av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
return -1;
}
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 -1;
}
while (bytestream_get_le32(&header) != MKTAG('f','m','t',' ')) {
len = bytestream_get_le32(&header);
if(len<0 || end - header - 8 < len)
return AVERROR_INVALIDDATA;
header += len;
}
len = bytestream_get_le32(&header);
if (len < 16) {
av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n");
return -1;
}
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 -1;
}
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 -1;
}
len -= 16;
if (len > 0)
av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len);
return 0;
}
static void interleave_buffer(int16_t *samples, int nchan, int blocksize,
int32_t **buffer)
{
int i, chan;
for (i=0; i<blocksize; i++)
for (chan=0; chan < nchan; chan++)
*samples++ = av_clip_int16(buffer[chan][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 -1;
}
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 > MAX_CHANNELS) {
av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
return -1;
}
s->avctx->channels = s->channels;
/* get blocksize if version > 0 */
if (s->version > 0) {
int skip_bytes, 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 -1;
}
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 -1;
}
for (i=0; i<s->header_size; i++)
s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
if (decode_wave_header(s->avctx, s->header, s->header_size) < 0)
return -1;
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)
{
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) {
int 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: {
int 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 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) {
/* get output buffer */
s->frame.nb_samples = s->blocksize;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
/* interleave output */
interleave_buffer((int16_t *)s->frame.data[0], s->channels,
s->blocksize, s->decoded);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
}
}
}
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 -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] = 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 = 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"),
};
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