Merge branch 'release/0.7' into oldabi

* release/0.7: (33 commits)
  Update for 0.7.8
  svq1dec: call avcodec_set_dimensions() after dimensions changed. Fixes NGS00148
  vp3dec: Check coefficient index in vp3_dequant() Fixes NGS00145
  qdm2dec: fix buffer overflow. Fixes NGS00144
  h264: Fix invalid interlaced progressive MB combinations for direct mode prediction. Fixes Ticket312
  mpegvideo: dont use ff_mspel_motion() for vc1 Fixes Ticket655
  imgutils: Fix illegal read.
  ac3probe: Detect Sonic Foundry Soft Encode AC3 as raw AC3. Our ac3 code chain can handle it fine. More ideal would be to write a demuxer that actually extracts what can be from the additional headers and uses it for whatever it can be used for.
  mjpeg: support mpo Fixes stereoscopic_photo.mpo
  Add a version bump and APIchanges entry for avcodec_open2 and avformat_find_stream_info.
  lavf: fix multiplication overflow in avformat_find_stream_info()
  lavf: fix invalid reads in avformat_find_stream_info()
  lavf: add avformat_find_stream_info()
  lavc: fix parentheses placement in avcodec_open2().
  lavc: introduce avcodec_open2() as a replacement for avcodec_open().
  rawdec: use a default sample rate if none is specified. Fixes "ffmpeg -f s16le -i /dev/zero"
  rawdec: add check on sample_rate
  qdm2dec: check remaining input bits in the mainloop of qdm2_fft_decode_tones() This is neccessary but likely not sufficient to prevent out of array reads.
  cinepak: check strip_size
  wma: Check channel number before init. Fixes Ticket240
  ...

Conflicts:
	RELEASE
	doc/APIchanges
	libavcodec/avcodec.h
	libavcodec/utils.c
	libavcodec/version.h
	libavdevice/v4l2.c
	libavformat/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 8 insertions, 3 deletions

diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index bc9ba95afa..e2eb0eacb8 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -77,6 +77,7 @@ do { \
 #define SAMPLES_NEEDED_2(why) \
      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
 
+#define QDM2_MAX_FRAME_SIZE 512
 
 typedef int8_t sb_int8_array[2][30][64];
 
@@ -169,7 +170,7 @@ typedef struct {
     /// I/O data
     const uint8_t *compressed_data;
     int compressed_size;
-    float output_buffer[1024];
+    float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
 
     /// Synthesis filter
     MPADSPContext mpadsp;
@@ -1328,7 +1329,7 @@ static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *
     local_int_10 = 1 << (q->group_order - duration - 1);
     offset = 1;
 
-    while (1) {
+    while (get_bits_left(gb)>0) {
         if (q->superblocktype_2_3) {
             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
                 offset = 1;
@@ -1823,7 +1824,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
     // something like max decodable tones
     s->group_order = av_log2(s->group_size) + 1;
     s->frame_size = s->group_size / 16; // 16 iterations per super block
-    if (s->frame_size > FF_ARRAY_ELEMS(s->output_buffer) / 2)
+
+    if (s->frame_size > QDM2_MAX_FRAME_SIZE)
         return AVERROR_INVALIDDATA;
 
     s->sub_sampling = s->fft_order - 7;
@@ -1894,6 +1896,9 @@ static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
     int ch, i;
     const int frame_size = (q->frame_size * q->channels);
 
+    if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
+        return -1;
+
     /* select input buffer */
     q->compressed_data = in;
     q->compressed_size = q->checksum_size;