Merge remote-tracking branch 'qatar/master'

* qatar/master: (58 commits)
  amrnbdec: check frame size before decoding.
  cscd: use negative error values to indicate decode_init() failures.
  h264: prevent overreads in intra PCM decoding.
  FATE: do not decode audio in the nuv test.
  dxa: set audio stream time base using the sample rate
  psx-str: do not allow seeking by bytes
  asfdec: Do not set AVCodecContext.frame_size
  vqf: set packet parameters after av_new_packet()
  mpegaudiodec: use DSPUtil.butterflies_float().
  FATE: add mp3 test for sample that exhibited false overreads
  fate: add cdxl test for bit line plane arrangement
  vmnc: return error on decode_init() failure.
  libvorbis: add/update error messages
  libvorbis: use AVFifoBuffer for output packet buffer
  libvorbis: remove unneeded e_o_s check
  libvorbis: check return values for functions that can return errors
  libvorbis: use float input instead of s16
  libvorbis: do not flush libvorbis analysis if dsp state was not initialized
  libvorbis: use VBR by default, with default quality of 3
  libvorbis: fix use of minrate/maxrate AVOptions
  ...

Conflicts:
	Changelog
	doc/APIchanges
	libavcodec/avcodec.h
	libavcodec/dpxenc.c
	libavcodec/libvorbis.c
	libavcodec/vmnc.c
	libavformat/asfdec.c
	libavformat/id3v2enc.c
	libavformat/internal.h
	libavformat/mp3enc.c
	libavformat/utils.c
	libavformat/version.h
	libswscale/utils.c
	tests/fate/video.mak
	tests/ref/fate/nuv
	tests/ref/fate/prores-alpha
	tests/ref/lavf/ffm
	tests/ref/vsynth1/prores
	tests/ref/vsynth2/prores

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

1 files changed, 5 insertions, 21 deletions

diff --git a/libavformat/asfdec.c b/libavformat/asfdec.c
index b37cbb04d3..cdec63b481 100644
--- a/libavformat/asfdec.c
+++ b/libavformat/asfdec.c
@@ -26,7 +26,6 @@
 #include "libavutil/avstring.h"
 #include "libavutil/dict.h"
 #include "libavutil/mathematics.h"
-#include "libavcodec/mpegaudio.h"
 #include "avformat.h"
 #include "internal.h"
 #include "avio_internal.h"
@@ -199,6 +198,8 @@ static int asf_read_file_properties(AVFormatContext *s, int64_t size)
     asf->hdr.flags              = avio_rl32(pb);
     asf->hdr.min_pktsize        = avio_rl32(pb);
     asf->hdr.max_pktsize        = avio_rl32(pb);
+    if (asf->hdr.min_pktsize >= (1U<<29))
+        return AVERROR_INVALIDDATA;
     asf->hdr.max_bitrate        = avio_rl32(pb);
     s->packet_size = asf->hdr.max_pktsize;
 
@@ -317,25 +318,6 @@ static int asf_read_stream_properties(AVFormatContext *s, int64_t size)
                     || asf_st->ds_packet_size % asf_st->ds_chunk_size)
                 asf_st->ds_span = 0; // disable descrambling
         }
-        switch (st->codec->codec_id) {
-            case CODEC_ID_MP3:
-                st->codec->frame_size = MPA_FRAME_SIZE;
-                break;
-            case CODEC_ID_PCM_S16LE:
-            case CODEC_ID_PCM_S16BE:
-            case CODEC_ID_PCM_U16LE:
-            case CODEC_ID_PCM_U16BE:
-            case CODEC_ID_PCM_S8:
-            case CODEC_ID_PCM_U8:
-            case CODEC_ID_PCM_ALAW:
-            case CODEC_ID_PCM_MULAW:
-                st->codec->frame_size = 1;
-                break;
-            default:
-                /* This is probably wrong, but it prevents a crash later */
-                st->codec->frame_size = 1;
-                break;
-        }
     } else if (type == AVMEDIA_TYPE_VIDEO &&
             size - (avio_tell(pb) - pos1 + 24) >= 51) {
         avio_rl32(pb);
@@ -612,7 +594,9 @@ static int asf_read_header(AVFormatContext *s)
         if (gsize < 24)
             return -1;
         if (!ff_guidcmp(&g, &ff_asf_file_header)) {
-            asf_read_file_properties(s, gsize);
+            int ret = asf_read_file_properties(s, gsize);
+            if (ret < 0)
+                return ret;
         } else if (!ff_guidcmp(&g, &ff_asf_stream_header)) {
             asf_read_stream_properties(s, gsize);
         } else if (!ff_guidcmp(&g, &ff_asf_comment_header)) {