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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavformat/rmdec.c
Michael Niedermayer 268098d8b2 Merge remote-tracking branch 'qatar/master'
* qatar/master: (29 commits)
  amrwb: remove duplicate arguments from extrapolate_isf().
  amrwb: error out early if mode is invalid.
  h264: change underread for 10bit QPEL to overread.
  matroska: check buffer size for RM-style byte reordering.
  vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
  vp8: change int stride to ptrdiff_t stride.
  wma: fix invalid buffer size assumptions causing random overreads.
  Windows Media Audio Lossless decoder
  rv10/20: Fix slice overflow with checked bitstream reader.
  h263dec: Disallow width/height changing with frame threads.
  rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
  rmdec: Honor .RMF tag size rather than assuming 18.
  g722: Fix the QMF scaling
  r3d: don't set codec timebase.
  electronicarts: set timebase for tgv video.
  electronicarts: parse the framerate for cmv video.
  ogg: don't set codec timebase
  electronicarts: don't set codec timebase
  avs: don't set codec timebase
  wavpack: Fix an integer overflow
  ...

Conflicts:
	libavcodec/arm/vp8dsp_init_arm.c
	libavcodec/fraps.c
	libavcodec/h264.c
	libavcodec/mpeg4videodec.c
	libavcodec/mpegvideo.c
	libavcodec/msmpeg4.c
	libavcodec/pnmdec.c
	libavcodec/qpeg.c
	libavcodec/rawenc.c
	libavcodec/ulti.c
	libavcodec/vcr1.c
	libavcodec/version.h
	libavcodec/wmalosslessdec.c
	libavformat/electronicarts.c
	libswscale/ppc/yuv2rgb_altivec.c
	tests/ref/acodec/g722
	tests/ref/fate/ea-cmv

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-03 00:23:10 +01:00

993 lines
32 KiB
C

/*
* "Real" compatible demuxer.
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* 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
*/
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/dict.h"
#include "avformat.h"
#include "internal.h"
#include "riff.h"
#include "rm.h"
#define DEINT_ID_GENR MKTAG('g', 'e', 'n', 'r') ///< interleaving for Cooker/Atrac
#define DEINT_ID_INT0 MKTAG('I', 'n', 't', '0') ///< no interleaving needed
#define DEINT_ID_INT4 MKTAG('I', 'n', 't', '4') ///< interleaving for 28.8
#define DEINT_ID_SIPR MKTAG('s', 'i', 'p', 'r') ///< interleaving for Sipro
#define DEINT_ID_VBRF MKTAG('v', 'b', 'r', 'f') ///< VBR case for AAC
#define DEINT_ID_VBRS MKTAG('v', 'b', 'r', 's') ///< VBR case for AAC
struct RMStream {
AVPacket pkt; ///< place to store merged video frame / reordered audio data
int videobufsize; ///< current assembled frame size
int videobufpos; ///< position for the next slice in the video buffer
int curpic_num; ///< picture number of current frame
int cur_slice, slices;
int64_t pktpos; ///< first slice position in file
/// Audio descrambling matrix parameters
int64_t audiotimestamp; ///< Audio packet timestamp
int sub_packet_cnt; // Subpacket counter, used while reading
int sub_packet_size, sub_packet_h, coded_framesize; ///< Descrambling parameters from container
int audio_framesize; /// Audio frame size from container
int sub_packet_lengths[16]; /// Length of each subpacket
int32_t deint_id; ///< deinterleaver used in audio stream
};
typedef struct {
int nb_packets;
int old_format;
int current_stream;
int remaining_len;
int audio_stream_num; ///< Stream number for audio packets
int audio_pkt_cnt; ///< Output packet counter
} RMDemuxContext;
static const unsigned char sipr_swaps[38][2] = {
{ 0, 63 }, { 1, 22 }, { 2, 44 }, { 3, 90 },
{ 5, 81 }, { 7, 31 }, { 8, 86 }, { 9, 58 },
{ 10, 36 }, { 12, 68 }, { 13, 39 }, { 14, 73 },
{ 15, 53 }, { 16, 69 }, { 17, 57 }, { 19, 88 },
{ 20, 34 }, { 21, 71 }, { 24, 46 }, { 25, 94 },
{ 26, 54 }, { 28, 75 }, { 29, 50 }, { 32, 70 },
{ 33, 92 }, { 35, 74 }, { 38, 85 }, { 40, 56 },
{ 42, 87 }, { 43, 65 }, { 45, 59 }, { 48, 79 },
{ 49, 93 }, { 51, 89 }, { 55, 95 }, { 61, 76 },
{ 67, 83 }, { 77, 80 }
};
const unsigned char ff_sipr_subpk_size[4] = { 29, 19, 37, 20 };
static inline void get_strl(AVIOContext *pb, char *buf, int buf_size, int len)
{
int i;
char *q, r;
q = buf;
for(i=0;i<len;i++) {
r = avio_r8(pb);
if (i < buf_size - 1)
*q++ = r;
}
if (buf_size > 0) *q = '\0';
}
static void get_str8(AVIOContext *pb, char *buf, int buf_size)
{
get_strl(pb, buf, buf_size, avio_r8(pb));
}
static int rm_read_extradata(AVIOContext *pb, AVCodecContext *avctx, unsigned size)
{
if (size >= 1<<24)
return -1;
avctx->extradata = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = avio_read(pb, avctx->extradata, size);
memset(avctx->extradata + avctx->extradata_size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
if (avctx->extradata_size != size)
return AVERROR(EIO);
return 0;
}
static void rm_read_metadata(AVFormatContext *s, int wide)
{
char buf[1024];
int i;
for (i=0; i<FF_ARRAY_ELEMS(ff_rm_metadata); i++) {
int len = wide ? avio_rb16(s->pb) : avio_r8(s->pb);
get_strl(s->pb, buf, sizeof(buf), len);
av_dict_set(&s->metadata, ff_rm_metadata[i], buf, 0);
}
}
RMStream *ff_rm_alloc_rmstream (void)
{
RMStream *rms = av_mallocz(sizeof(RMStream));
rms->curpic_num = -1;
return rms;
}
void ff_rm_free_rmstream (RMStream *rms)
{
av_free_packet(&rms->pkt);
}
static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
AVStream *st, RMStream *ast, int read_all)
{
char buf[256];
uint32_t version;
int ret;
/* ra type header */
version = avio_rb16(pb); /* version */
if (version == 3) {
int header_size = avio_rb16(pb);
int64_t startpos = avio_tell(pb);
avio_skip(pb, 14);
rm_read_metadata(s, 0);
if ((startpos + header_size) >= avio_tell(pb) + 2) {
// fourcc (should always be "lpcJ")
avio_r8(pb);
get_str8(pb, buf, sizeof(buf));
}
// Skip extra header crap (this should never happen)
if ((startpos + header_size) > avio_tell(pb))
avio_skip(pb, header_size + startpos - avio_tell(pb));
st->codec->sample_rate = 8000;
st->codec->channels = 1;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = CODEC_ID_RA_144;
ast->deint_id = DEINT_ID_INT0;
} else {
int flavor, sub_packet_h, coded_framesize, sub_packet_size;
int codecdata_length;
/* old version (4) */
avio_skip(pb, 2); /* unused */
avio_rb32(pb); /* .ra4 */
avio_rb32(pb); /* data size */
avio_rb16(pb); /* version2 */
avio_rb32(pb); /* header size */
flavor= avio_rb16(pb); /* add codec info / flavor */
ast->coded_framesize = coded_framesize = avio_rb32(pb); /* coded frame size */
avio_rb32(pb); /* ??? */
avio_rb32(pb); /* ??? */
avio_rb32(pb); /* ??? */
ast->sub_packet_h = sub_packet_h = avio_rb16(pb); /* 1 */
st->codec->block_align= avio_rb16(pb); /* frame size */
ast->sub_packet_size = sub_packet_size = avio_rb16(pb); /* sub packet size */
avio_rb16(pb); /* ??? */
if (version == 5) {
avio_rb16(pb); avio_rb16(pb); avio_rb16(pb);
}
st->codec->sample_rate = avio_rb16(pb);
avio_rb32(pb);
st->codec->channels = avio_rb16(pb);
if (version == 5) {
ast->deint_id = avio_rl32(pb);
avio_read(pb, buf, 4);
buf[4] = 0;
} else {
get_str8(pb, buf, sizeof(buf)); /* desc */
ast->deint_id = AV_RL32(buf);
get_str8(pb, buf, sizeof(buf)); /* desc */
}
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_tag = AV_RL32(buf);
st->codec->codec_id = ff_codec_get_id(ff_rm_codec_tags,
st->codec->codec_tag);
switch (st->codec->codec_id) {
case CODEC_ID_AC3:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
case CODEC_ID_RA_288:
st->codec->extradata_size= 0;
ast->audio_framesize = st->codec->block_align;
st->codec->block_align = coded_framesize;
break;
case CODEC_ID_COOK:
case CODEC_ID_ATRAC3:
case CODEC_ID_SIPR:
avio_rb16(pb); avio_r8(pb);
if (version == 5)
avio_r8(pb);
codecdata_length = avio_rb32(pb);
if(codecdata_length + FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)codecdata_length){
av_log(s, AV_LOG_ERROR, "codecdata_length too large\n");
return -1;
}
ast->audio_framesize = st->codec->block_align;
if (st->codec->codec_id == CODEC_ID_SIPR) {
if (flavor > 3) {
av_log(s, AV_LOG_ERROR, "bad SIPR file flavor %d\n",
flavor);
return -1;
}
st->codec->block_align = ff_sipr_subpk_size[flavor];
} else {
if(sub_packet_size <= 0){
av_log(s, AV_LOG_ERROR, "sub_packet_size is invalid\n");
return -1;
}
st->codec->block_align = ast->sub_packet_size;
}
if ((ret = rm_read_extradata(pb, st->codec, codecdata_length)) < 0)
return ret;
break;
case CODEC_ID_AAC:
avio_rb16(pb); avio_r8(pb);
if (version == 5)
avio_r8(pb);
codecdata_length = avio_rb32(pb);
if(codecdata_length + FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)codecdata_length){
av_log(s, AV_LOG_ERROR, "codecdata_length too large\n");
return -1;
}
if (codecdata_length >= 1) {
avio_r8(pb);
if ((ret = rm_read_extradata(pb, st->codec, codecdata_length - 1)) < 0)
return ret;
}
break;
default:
av_strlcpy(st->codec->codec_name, buf, sizeof(st->codec->codec_name));
}
if (ast->deint_id == DEINT_ID_INT4 ||
ast->deint_id == DEINT_ID_GENR ||
ast->deint_id == DEINT_ID_SIPR) {
if (st->codec->block_align <= 0 ||
ast->audio_framesize * sub_packet_h > (unsigned)INT_MAX ||
ast->audio_framesize * sub_packet_h < st->codec->block_align)
return AVERROR_INVALIDDATA;
if (av_new_packet(&ast->pkt, ast->audio_framesize * sub_packet_h) < 0)
return AVERROR(ENOMEM);
}
switch (ast->deint_id) {
case DEINT_ID_INT4:
if (ast->coded_framesize > ast->audio_framesize ||
sub_packet_h <= 1 ||
ast->coded_framesize * sub_packet_h > (2 + (sub_packet_h & 1)) * ast->audio_framesize)
return AVERROR_INVALIDDATA;
break;
case DEINT_ID_GENR:
if (ast->sub_packet_size <= 0 ||
ast->sub_packet_size > ast->audio_framesize)
return AVERROR_INVALIDDATA;
break;
case DEINT_ID_SIPR:
case DEINT_ID_INT0:
case DEINT_ID_VBRS:
case DEINT_ID_VBRF:
break;
default:
av_log(NULL,0,"Unknown interleaver %X\n", ast->deint_id);
return AVERROR_INVALIDDATA;
}
if (read_all) {
avio_r8(pb);
avio_r8(pb);
avio_r8(pb);
rm_read_metadata(s, 0);
}
}
return 0;
}
int
ff_rm_read_mdpr_codecdata (AVFormatContext *s, AVIOContext *pb,
AVStream *st, RMStream *rst, int codec_data_size)
{
unsigned int v;
int size;
int64_t codec_pos;
int ret;
avpriv_set_pts_info(st, 64, 1, 1000);
codec_pos = avio_tell(pb);
v = avio_rb32(pb);
if (v == MKTAG(0xfd, 'a', 'r', '.')) {
/* ra type header */
if (rm_read_audio_stream_info(s, pb, st, rst, 0))
return -1;
} else {
int fps;
if (avio_rl32(pb) != MKTAG('V', 'I', 'D', 'O')) {
fail1:
av_log(s, AV_LOG_WARNING, "Unsupported stream type %08x\n", v);
goto skip;
}
st->codec->codec_tag = avio_rl32(pb);
st->codec->codec_id = ff_codec_get_id(ff_rm_codec_tags,
st->codec->codec_tag);
// av_log(s, AV_LOG_DEBUG, "%X %X\n", st->codec->codec_tag, MKTAG('R', 'V', '2', '0'));
if (st->codec->codec_id == CODEC_ID_NONE)
goto fail1;
st->codec->width = avio_rb16(pb);
st->codec->height = avio_rb16(pb);
avio_skip(pb, 2); // looks like bits per sample
avio_skip(pb, 4); // always zero?
st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
st->need_parsing = AVSTREAM_PARSE_TIMESTAMPS;
fps = avio_rb32(pb);
if ((ret = rm_read_extradata(pb, st->codec, codec_data_size - (avio_tell(pb) - codec_pos))) < 0)
return ret;
av_reduce(&st->r_frame_rate.den, &st->r_frame_rate.num,
0x10000, fps, (1 << 30) - 1);
st->avg_frame_rate = st->r_frame_rate;
}
skip:
/* skip codec info */
size = avio_tell(pb) - codec_pos;
avio_skip(pb, codec_data_size - size);
return 0;
}
/** this function assumes that the demuxer has already seeked to the start
* of the INDX chunk, and will bail out if not. */
static int rm_read_index(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
unsigned int size, n_pkts, str_id, next_off, n, pos, pts;
AVStream *st;
do {
if (avio_rl32(pb) != MKTAG('I','N','D','X'))
return -1;
size = avio_rb32(pb);
if (size < 20)
return -1;
avio_skip(pb, 2);
n_pkts = avio_rb32(pb);
str_id = avio_rb16(pb);
next_off = avio_rb32(pb);
for (n = 0; n < s->nb_streams; n++)
if (s->streams[n]->id == str_id) {
st = s->streams[n];
break;
}
if (n == s->nb_streams) {
av_log(s, AV_LOG_ERROR,
"Invalid stream index %d for index at pos %"PRId64"\n",
str_id, avio_tell(pb));
goto skip;
} else if ((avio_size(pb) - avio_tell(pb)) / 14 < n_pkts) {
av_log(s, AV_LOG_ERROR,
"Nr. of packets in packet index for stream index %d "
"exceeds filesize (%"PRId64" at %"PRId64" = %"PRId64")\n",
str_id, avio_size(pb), avio_tell(pb),
(avio_size(pb) - avio_tell(pb)) / 14);
goto skip;
}
for (n = 0; n < n_pkts; n++) {
avio_skip(pb, 2);
pts = avio_rb32(pb);
pos = avio_rb32(pb);
avio_skip(pb, 4); /* packet no. */
av_add_index_entry(st, pos, pts, 0, 0, AVINDEX_KEYFRAME);
}
skip:
if (next_off && avio_tell(pb) < next_off &&
avio_seek(pb, next_off, SEEK_SET) < 0) {
av_log(s, AV_LOG_ERROR,
"Non-linear index detected, not supported\n");
return -1;
}
} while (next_off);
return 0;
}
static int rm_read_header_old(AVFormatContext *s)
{
RMDemuxContext *rm = s->priv_data;
AVStream *st;
rm->old_format = 1;
st = avformat_new_stream(s, NULL);
if (!st)
return -1;
st->priv_data = ff_rm_alloc_rmstream();
return rm_read_audio_stream_info(s, s->pb, st, st->priv_data, 1);
}
static int rm_read_header(AVFormatContext *s)
{
RMDemuxContext *rm = s->priv_data;
AVStream *st;
AVIOContext *pb = s->pb;
unsigned int tag;
int tag_size;
unsigned int start_time, duration;
unsigned int data_off = 0, indx_off = 0;
char buf[128];
int flags = 0;
tag = avio_rl32(pb);
if (tag == MKTAG('.', 'r', 'a', 0xfd)) {
/* very old .ra format */
return rm_read_header_old(s);
} else if (tag != MKTAG('.', 'R', 'M', 'F')) {
return AVERROR(EIO);
}
tag_size = avio_rb32(pb);
avio_skip(pb, tag_size - 8);
for(;;) {
if (url_feof(pb))
return -1;
tag = avio_rl32(pb);
tag_size = avio_rb32(pb);
avio_rb16(pb);
av_dlog(s, "tag=%c%c%c%c (%08x) size=%d\n",
(tag ) & 0xff,
(tag >> 8) & 0xff,
(tag >> 16) & 0xff,
(tag >> 24) & 0xff,
tag,
tag_size);
if (tag_size < 10 && tag != MKTAG('D', 'A', 'T', 'A'))
return -1;
switch(tag) {
case MKTAG('P', 'R', 'O', 'P'):
/* file header */
avio_rb32(pb); /* max bit rate */
avio_rb32(pb); /* avg bit rate */
avio_rb32(pb); /* max packet size */
avio_rb32(pb); /* avg packet size */
avio_rb32(pb); /* nb packets */
avio_rb32(pb); /* duration */
avio_rb32(pb); /* preroll */
indx_off = avio_rb32(pb); /* index offset */
data_off = avio_rb32(pb); /* data offset */
avio_rb16(pb); /* nb streams */
flags = avio_rb16(pb); /* flags */
break;
case MKTAG('C', 'O', 'N', 'T'):
rm_read_metadata(s, 1);
break;
case MKTAG('M', 'D', 'P', 'R'):
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->id = avio_rb16(pb);
avio_rb32(pb); /* max bit rate */
st->codec->bit_rate = avio_rb32(pb); /* bit rate */
avio_rb32(pb); /* max packet size */
avio_rb32(pb); /* avg packet size */
start_time = avio_rb32(pb); /* start time */
avio_rb32(pb); /* preroll */
duration = avio_rb32(pb); /* duration */
st->start_time = start_time;
st->duration = duration;
get_str8(pb, buf, sizeof(buf)); /* desc */
get_str8(pb, buf, sizeof(buf)); /* mimetype */
st->codec->codec_type = AVMEDIA_TYPE_DATA;
st->priv_data = ff_rm_alloc_rmstream();
if (ff_rm_read_mdpr_codecdata(s, s->pb, st, st->priv_data,
avio_rb32(pb)) < 0)
return -1;
break;
case MKTAG('D', 'A', 'T', 'A'):
goto header_end;
default:
/* unknown tag: skip it */
avio_skip(pb, tag_size - 10);
break;
}
}
header_end:
rm->nb_packets = avio_rb32(pb); /* number of packets */
if (!rm->nb_packets && (flags & 4))
rm->nb_packets = 3600 * 25;
avio_rb32(pb); /* next data header */
if (!data_off)
data_off = avio_tell(pb) - 18;
if (indx_off && pb->seekable && !(s->flags & AVFMT_FLAG_IGNIDX) &&
avio_seek(pb, indx_off, SEEK_SET) >= 0) {
rm_read_index(s);
avio_seek(pb, data_off + 18, SEEK_SET);
}
return 0;
}
static int get_num(AVIOContext *pb, int *len)
{
int n, n1;
n = avio_rb16(pb);
(*len)-=2;
n &= 0x7FFF;
if (n >= 0x4000) {
return n - 0x4000;
} else {
n1 = avio_rb16(pb);
(*len)-=2;
return (n << 16) | n1;
}
}
/* multiple of 20 bytes for ra144 (ugly) */
#define RAW_PACKET_SIZE 1000
static int sync(AVFormatContext *s, int64_t *timestamp, int *flags, int *stream_index, int64_t *pos){
RMDemuxContext *rm = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st;
uint32_t state=0xFFFFFFFF;
while(!url_feof(pb)){
int len, num, i;
*pos= avio_tell(pb) - 3;
if(rm->remaining_len > 0){
num= rm->current_stream;
len= rm->remaining_len;
*timestamp = AV_NOPTS_VALUE;
*flags= 0;
}else{
state= (state<<8) + avio_r8(pb);
if(state == MKBETAG('I', 'N', 'D', 'X')){
int n_pkts, expected_len;
len = avio_rb32(pb);
avio_skip(pb, 2);
n_pkts = avio_rb32(pb);
expected_len = 20 + n_pkts * 14;
if (len == 20)
/* some files don't add index entries to chunk size... */
len = expected_len;
else if (len != expected_len)
av_log(s, AV_LOG_WARNING,
"Index size %d (%d pkts) is wrong, should be %d.\n",
len, n_pkts, expected_len);
len -= 14; // we already read part of the index header
if(len<0)
continue;
goto skip;
} else if (state == MKBETAG('D','A','T','A')) {
av_log(s, AV_LOG_WARNING,
"DATA tag in middle of chunk, file may be broken.\n");
}
if(state > (unsigned)0xFFFF || state <= 12)
continue;
len=state - 12;
state= 0xFFFFFFFF;
num = avio_rb16(pb);
*timestamp = avio_rb32(pb);
avio_r8(pb); /* reserved */
*flags = avio_r8(pb); /* flags */
}
for(i=0;i<s->nb_streams;i++) {
st = s->streams[i];
if (num == st->id)
break;
}
if (i == s->nb_streams) {
skip:
/* skip packet if unknown number */
avio_skip(pb, len);
rm->remaining_len = 0;
continue;
}
*stream_index= i;
return len;
}
return -1;
}
static int rm_assemble_video_frame(AVFormatContext *s, AVIOContext *pb,
RMDemuxContext *rm, RMStream *vst,
AVPacket *pkt, int len, int *pseq,
int64_t *timestamp)
{
int hdr, seq, pic_num, len2, pos;
int type;
hdr = avio_r8(pb); len--;
type = hdr >> 6;
if(type != 3){ // not frame as a part of packet
seq = avio_r8(pb); len--;
}
if(type != 1){ // not whole frame
len2 = get_num(pb, &len);
pos = get_num(pb, &len);
pic_num = avio_r8(pb); len--;
}
if(len<0)
return -1;
rm->remaining_len = len;
if(type&1){ // frame, not slice
if(type == 3){ // frame as a part of packet
len= len2;
*timestamp = pos;
}
if(rm->remaining_len < len)
return -1;
rm->remaining_len -= len;
if(av_new_packet(pkt, len + 9) < 0)
return AVERROR(EIO);
pkt->data[0] = 0;
AV_WL32(pkt->data + 1, 1);
AV_WL32(pkt->data + 5, 0);
avio_read(pb, pkt->data + 9, len);
return 0;
}
//now we have to deal with single slice
*pseq = seq;
if((seq & 0x7F) == 1 || vst->curpic_num != pic_num){
vst->slices = ((hdr & 0x3F) << 1) + 1;
vst->videobufsize = len2 + 8*vst->slices + 1;
av_free_packet(&vst->pkt); //FIXME this should be output.
if(av_new_packet(&vst->pkt, vst->videobufsize) < 0)
return AVERROR(ENOMEM);
vst->videobufpos = 8*vst->slices + 1;
vst->cur_slice = 0;
vst->curpic_num = pic_num;
vst->pktpos = avio_tell(pb);
}
if(type == 2)
len = FFMIN(len, pos);
if(++vst->cur_slice > vst->slices)
return 1;
AV_WL32(vst->pkt.data - 7 + 8*vst->cur_slice, 1);
AV_WL32(vst->pkt.data - 3 + 8*vst->cur_slice, vst->videobufpos - 8*vst->slices - 1);
if(vst->videobufpos + len > vst->videobufsize)
return 1;
if (avio_read(pb, vst->pkt.data + vst->videobufpos, len) != len)
return AVERROR(EIO);
vst->videobufpos += len;
rm->remaining_len-= len;
if (type == 2 || vst->videobufpos == vst->videobufsize) {
vst->pkt.data[0] = vst->cur_slice-1;
*pkt= vst->pkt;
vst->pkt.data= NULL;
vst->pkt.size= 0;
if(vst->slices != vst->cur_slice) //FIXME find out how to set slices correct from the begin
memmove(pkt->data + 1 + 8*vst->cur_slice, pkt->data + 1 + 8*vst->slices,
vst->videobufpos - 1 - 8*vst->slices);
pkt->size = vst->videobufpos + 8*(vst->cur_slice - vst->slices);
pkt->pts = AV_NOPTS_VALUE;
pkt->pos = vst->pktpos;
vst->slices = 0;
return 0;
}
return 1;
}
static inline void
rm_ac3_swap_bytes (AVStream *st, AVPacket *pkt)
{
uint8_t *ptr;
int j;
if (st->codec->codec_id == CODEC_ID_AC3) {
ptr = pkt->data;
for (j=0;j<pkt->size;j+=2) {
FFSWAP(int, ptr[0], ptr[1]);
ptr += 2;
}
}
}
/**
* Perform 4-bit block reordering for SIPR data.
* @todo This can be optimized, e.g. use memcpy() if data blocks are aligned
*/
void ff_rm_reorder_sipr_data(uint8_t *buf, int sub_packet_h, int framesize)
{
int n, bs = sub_packet_h * framesize * 2 / 96; // nibbles per subpacket
for (n = 0; n < 38; n++) {
int j;
int i = bs * sipr_swaps[n][0];
int o = bs * sipr_swaps[n][1];
/* swap 4bit-nibbles of block 'i' with 'o' */
for (j = 0; j < bs; j++, i++, o++) {
int x = (buf[i >> 1] >> (4 * (i & 1))) & 0xF,
y = (buf[o >> 1] >> (4 * (o & 1))) & 0xF;
buf[o >> 1] = (x << (4 * (o & 1))) |
(buf[o >> 1] & (0xF << (4 * !(o & 1))));
buf[i >> 1] = (y << (4 * (i & 1))) |
(buf[i >> 1] & (0xF << (4 * !(i & 1))));
}
}
}
int
ff_rm_parse_packet (AVFormatContext *s, AVIOContext *pb,
AVStream *st, RMStream *ast, int len, AVPacket *pkt,
int *seq, int flags, int64_t timestamp)
{
RMDemuxContext *rm = s->priv_data;
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
rm->current_stream= st->id;
if(rm_assemble_video_frame(s, pb, rm, ast, pkt, len, seq, &timestamp))
return -1; //got partial frame
} else if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if ((ast->deint_id == DEINT_ID_GENR) ||
(ast->deint_id == DEINT_ID_INT4) ||
(ast->deint_id == DEINT_ID_SIPR)) {
int x;
int sps = ast->sub_packet_size;
int cfs = ast->coded_framesize;
int h = ast->sub_packet_h;
int y = ast->sub_packet_cnt;
int w = ast->audio_framesize;
if (flags & 2)
y = ast->sub_packet_cnt = 0;
if (!y)
ast->audiotimestamp = timestamp;
switch (ast->deint_id) {
case DEINT_ID_INT4:
for (x = 0; x < h/2; x++)
avio_read(pb, ast->pkt.data+x*2*w+y*cfs, cfs);
break;
case DEINT_ID_GENR:
for (x = 0; x < w/sps; x++)
avio_read(pb, ast->pkt.data+sps*(h*x+((h+1)/2)*(y&1)+(y>>1)), sps);
break;
case DEINT_ID_SIPR:
avio_read(pb, ast->pkt.data + y * w, w);
break;
}
if (++(ast->sub_packet_cnt) < h)
return -1;
if (ast->deint_id == DEINT_ID_SIPR)
ff_rm_reorder_sipr_data(ast->pkt.data, h, w);
ast->sub_packet_cnt = 0;
rm->audio_stream_num = st->index;
rm->audio_pkt_cnt = h * w / st->codec->block_align;
} else if ((ast->deint_id == DEINT_ID_VBRF) ||
(ast->deint_id == DEINT_ID_VBRS)) {
int x;
rm->audio_stream_num = st->index;
ast->sub_packet_cnt = (avio_rb16(pb) & 0xf0) >> 4;
if (ast->sub_packet_cnt) {
for (x = 0; x < ast->sub_packet_cnt; x++)
ast->sub_packet_lengths[x] = avio_rb16(pb);
rm->audio_pkt_cnt = ast->sub_packet_cnt;
ast->audiotimestamp = timestamp;
} else
return -1;
} else {
av_get_packet(pb, pkt, len);
rm_ac3_swap_bytes(st, pkt);
}
} else
av_get_packet(pb, pkt, len);
pkt->stream_index = st->index;
#if 0
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
if(st->codec->codec_id == CODEC_ID_RV20){
int seq= 128*(pkt->data[2]&0x7F) + (pkt->data[3]>>1);
av_log(s, AV_LOG_DEBUG, "%d %"PRId64" %d\n", *timestamp, *timestamp*512LL/25, seq);
seq |= (timestamp&~0x3FFF);
if(seq - timestamp > 0x2000) seq -= 0x4000;
if(seq - timestamp < -0x2000) seq += 0x4000;
}
}
#endif
pkt->pts = timestamp;
if (flags & 2)
pkt->flags |= AV_PKT_FLAG_KEY;
return st->codec->codec_type == AVMEDIA_TYPE_AUDIO ? rm->audio_pkt_cnt : 0;
}
int
ff_rm_retrieve_cache (AVFormatContext *s, AVIOContext *pb,
AVStream *st, RMStream *ast, AVPacket *pkt)
{
RMDemuxContext *rm = s->priv_data;
assert (rm->audio_pkt_cnt > 0);
if (ast->deint_id == DEINT_ID_VBRF ||
ast->deint_id == DEINT_ID_VBRS)
av_get_packet(pb, pkt, ast->sub_packet_lengths[ast->sub_packet_cnt - rm->audio_pkt_cnt]);
else {
av_new_packet(pkt, st->codec->block_align);
memcpy(pkt->data, ast->pkt.data + st->codec->block_align * //FIXME avoid this
(ast->sub_packet_h * ast->audio_framesize / st->codec->block_align - rm->audio_pkt_cnt),
st->codec->block_align);
}
rm->audio_pkt_cnt--;
if ((pkt->pts = ast->audiotimestamp) != AV_NOPTS_VALUE) {
ast->audiotimestamp = AV_NOPTS_VALUE;
pkt->flags = AV_PKT_FLAG_KEY;
} else
pkt->flags = 0;
pkt->stream_index = st->index;
return rm->audio_pkt_cnt;
}
static int rm_read_packet(AVFormatContext *s, AVPacket *pkt)
{
RMDemuxContext *rm = s->priv_data;
AVStream *st;
int i, len, res, seq = 1;
int64_t timestamp, pos;
int flags;
for (;;) {
if (rm->audio_pkt_cnt) {
// If there are queued audio packet return them first
st = s->streams[rm->audio_stream_num];
ff_rm_retrieve_cache(s, s->pb, st, st->priv_data, pkt);
flags = 0;
} else {
if (rm->old_format) {
RMStream *ast;
st = s->streams[0];
ast = st->priv_data;
timestamp = AV_NOPTS_VALUE;
len = !ast->audio_framesize ? RAW_PACKET_SIZE :
ast->coded_framesize * ast->sub_packet_h / 2;
flags = (seq++ == 1) ? 2 : 0;
pos = avio_tell(s->pb);
} else {
len=sync(s, &timestamp, &flags, &i, &pos);
if (len > 0)
st = s->streams[i];
}
if(len<0 || url_feof(s->pb))
return AVERROR(EIO);
res = ff_rm_parse_packet (s, s->pb, st, st->priv_data, len, pkt,
&seq, flags, timestamp);
if((flags&2) && (seq&0x7F) == 1)
av_add_index_entry(st, pos, timestamp, 0, 0, AVINDEX_KEYFRAME);
if (res)
continue;
}
if( (st->discard >= AVDISCARD_NONKEY && !(flags&2))
|| st->discard >= AVDISCARD_ALL){
av_free_packet(pkt);
} else
break;
}
return 0;
}
static int rm_read_close(AVFormatContext *s)
{
int i;
for (i=0;i<s->nb_streams;i++)
ff_rm_free_rmstream(s->streams[i]->priv_data);
return 0;
}
static int rm_probe(AVProbeData *p)
{
/* check file header */
if ((p->buf[0] == '.' && p->buf[1] == 'R' &&
p->buf[2] == 'M' && p->buf[3] == 'F' &&
p->buf[4] == 0 && p->buf[5] == 0) ||
(p->buf[0] == '.' && p->buf[1] == 'r' &&
p->buf[2] == 'a' && p->buf[3] == 0xfd))
return AVPROBE_SCORE_MAX;
else
return 0;
}
static int64_t rm_read_dts(AVFormatContext *s, int stream_index,
int64_t *ppos, int64_t pos_limit)
{
RMDemuxContext *rm = s->priv_data;
int64_t pos, dts;
int stream_index2, flags, len, h;
pos = *ppos;
if(rm->old_format)
return AV_NOPTS_VALUE;
if (avio_seek(s->pb, pos, SEEK_SET) < 0)
return AV_NOPTS_VALUE;
rm->remaining_len=0;
for(;;){
int seq=1;
AVStream *st;
len=sync(s, &dts, &flags, &stream_index2, &pos);
if(len<0)
return AV_NOPTS_VALUE;
st = s->streams[stream_index2];
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
h= avio_r8(s->pb); len--;
if(!(h & 0x40)){
seq = avio_r8(s->pb); len--;
}
}
if((flags&2) && (seq&0x7F) == 1){
// av_log(s, AV_LOG_DEBUG, "%d %d-%d %"PRId64" %d\n", flags, stream_index2, stream_index, dts, seq);
av_add_index_entry(st, pos, dts, 0, 0, AVINDEX_KEYFRAME);
if(stream_index2 == stream_index)
break;
}
avio_skip(s->pb, len);
}
*ppos = pos;
return dts;
}
AVInputFormat ff_rm_demuxer = {
.name = "rm",
.long_name = NULL_IF_CONFIG_SMALL("RealMedia format"),
.priv_data_size = sizeof(RMDemuxContext),
.read_probe = rm_probe,
.read_header = rm_read_header,
.read_packet = rm_read_packet,
.read_close = rm_read_close,
.read_timestamp = rm_read_dts,
};
AVInputFormat ff_rdt_demuxer = {
.name = "rdt",
.long_name = NULL_IF_CONFIG_SMALL("RDT demuxer"),
.priv_data_size = sizeof(RMDemuxContext),
.read_close = rm_read_close,
.flags = AVFMT_NOFILE,
};