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authorMichael Niedermayer <michaelni@gmx.at>2011-09-22 01:41:55 +0200
committerMichael Niedermayer <michaelni@gmx.at>2011-09-22 01:48:45 +0200
commita99a35c8ea633a252d3a52a21691478f1f8c1763 (patch)
treee38f9dd8ff73ba17809e7475c08e41ffb5e99f78 /libavformat/rmdec.c
parenta582b028a416a29e60b4748b2ae4221ada11c2b8 (diff)
parent056e9efc8e06faed3516b610d8a74484f24ac07f (diff)
downloadffmpeg-a99a35c8ea633a252d3a52a21691478f1f8c1763.tar.gz
Merge branch 'release/0.8' into release/0.7
* release/0.8: (154 commits) vp6: partially propagate huffman tree building errors during coeff model parsing and fix misspelling Check for huffman tree building error in vp6 decoder. Release old pictures after a resolution change in vp5/6 decoder Check for missing reference in vp5/6 decoder. Check for invalid slices offsets in RV30/40 decoder. Check output buffer size in nellymoser decoder. Hack around gcc 4.6 breaking asm using call. Fix dxva2 decoding for some H264 samples. mp3demux: pass on error code on packet read. Check for invalid slice offsets in real decoder. rmdec: Reject invalid deinterleaving parameters Use deinterleavers for demangling audio packets in RealMedia. rv10: Reject slices that does not have the same type as the first one rmdec: use the deinterleaving mode and not the codec when creating audio packets. MAINTAINERS: add my GPG fingerprint. (cherry picked from commit 7882dc10f871bf25a848fe62a152f63814f9c7d1) Support 3IVD in isom, produced by 3ivx DivX Doctor. mpegpsdec: fix reading first mpegps packet (cherry picked from commit b2f230e23dd61112ac090b0c059d87b5f6bcb307) Avoid NULL dereference on corrupted bitstream with real decoder. Reject slices that does not have the same type than the first one in RV10/RV20 decoder. check all svq3_get_ue_golomb() returns. ... Conflicts: Doxyfile RELEASE VERSION libavcodec/rv34.c libavformat/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rmdec.c')
-rw-r--r--libavformat/rmdec.c86
1 files changed, 58 insertions, 28 deletions
diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
index c2eee815c2..708187e16f 100644
--- a/libavformat/rmdec.c
+++ b/libavformat/rmdec.c
@@ -26,6 +26,13 @@
#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
@@ -39,6 +46,7 @@ struct RMStream {
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 {
@@ -147,6 +155,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
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;
@@ -172,17 +181,19 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
avio_rb32(pb);
st->codec->channels = avio_rb16(pb);
if (version == 5) {
- avio_rb32(pb);
+ 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;
@@ -191,13 +202,6 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
st->codec->extradata_size= 0;
ast->audio_framesize = st->codec->block_align;
st->codec->block_align = coded_framesize;
-
- if(ast->audio_framesize >= UINT_MAX / sub_packet_h){
- av_log(s, AV_LOG_ERROR, "ast->audio_framesize * sub_packet_h too large\n");
- return -1;
- }
-
- av_new_packet(&ast->pkt, ast->audio_framesize * sub_packet_h);
break;
case CODEC_ID_COOK:
case CODEC_ID_ATRAC3:
@@ -228,13 +232,6 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
}
if ((ret = rm_read_extradata(pb, st->codec, codecdata_length)) < 0)
return ret;
-
- if(ast->audio_framesize >= UINT_MAX / sub_packet_h){
- av_log(s, AV_LOG_ERROR, "rm->audio_framesize * sub_packet_h too large\n");
- return -1;
- }
-
- av_new_packet(&ast->pkt, ast->audio_framesize * sub_packet_h);
break;
case CODEC_ID_AAC:
avio_rb16(pb); avio_r8(pb);
@@ -254,6 +251,37 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
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 ||
+ 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);
@@ -712,10 +740,9 @@ ff_rm_parse_packet (AVFormatContext *s, AVIOContext *pb,
if(rm_assemble_video_frame(s, pb, rm, ast, pkt, len, seq))
return -1; //got partial frame
} else if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- if ((st->codec->codec_id == CODEC_ID_RA_288) ||
- (st->codec->codec_id == CODEC_ID_COOK) ||
- (st->codec->codec_id == CODEC_ID_ATRAC3) ||
- (st->codec->codec_id == CODEC_ID_SIPR)) {
+ 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;
@@ -728,30 +755,30 @@ ff_rm_parse_packet (AVFormatContext *s, AVIOContext *pb,
if (!y)
ast->audiotimestamp = timestamp;
- switch(st->codec->codec_id) {
- case CODEC_ID_RA_288:
+ 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 CODEC_ID_ATRAC3:
- case CODEC_ID_COOK:
+ 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 CODEC_ID_SIPR:
+ case DEINT_ID_SIPR:
avio_read(pb, ast->pkt.data + y * w, w);
break;
}
if (++(ast->sub_packet_cnt) < h)
return -1;
- if (st->codec->codec_id == CODEC_ID_SIPR)
+ 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 (st->codec->codec_id == CODEC_ID_AAC) {
+ } 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;
@@ -799,7 +826,8 @@ ff_rm_retrieve_cache (AVFormatContext *s, AVIOContext *pb,
assert (rm->audio_pkt_cnt > 0);
- if (st->codec->codec_id == CODEC_ID_AAC)
+ 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);
@@ -905,7 +933,9 @@ static int64_t rm_read_dts(AVFormatContext *s, int stream_index,
if(rm->old_format)
return AV_NOPTS_VALUE;
- avio_seek(s->pb, pos, SEEK_SET);
+ if (avio_seek(s->pb, pos, SEEK_SET) < 0)
+ return AV_NOPTS_VALUE;
+
rm->remaining_len=0;
for(;;){
int seq=1;