correct AUDIO strf parsing patch by (Roman Shaposhnick <rvs at sun dot com>)

Originally committed as revision 1664 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/libavformat/asf.c b/libavformat/asf.c
index 5edde52..b9cd71e 100644 (file)
--- a/libavformat/asf.c
+++ b/libavformat/asf.c
@@ -806,7 +806,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
            asf->packet_size = asf->hdr.max_pktsize;
             asf->nb_packets = asf->hdr.packets_count;
         } else if (!memcmp(&g, &stream_header, sizeof(GUID))) {
-            int type, total_size;
+            int type, total_size, type_specific_size;
             unsigned int tag1;
             int64_t pos1, pos2;
 
@@ -832,7 +832,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
             }
             get_guid(pb, &g);
             total_size = get_le64(pb);
-            get_le32(pb);
+            type_specific_size = get_le32(pb);
             get_le32(pb);
            st->id = get_le16(pb) & 0x7f; /* stream id */
             // mapping of asf ID to AV stream ID;
@@ -842,7 +842,7 @@ static int asf_read_header(AVFormatContext *s, AVFormatParameters *ap)
            st->codec.codec_type = type;
             st->codec.frame_rate = 15 * s->pts_den / s->pts_num; // 15 fps default
             if (type == CODEC_TYPE_AUDIO) {
-                get_wav_header(pb, &st->codec, 1);
+                get_wav_header(pb, &st->codec, type_specific_size);
                /* We have to init the frame size at some point .... */
                pos2 = url_ftell(pb);
                if (gsize > (pos2 + 8 - pos1 + 24)) {

diff --git a/libavformat/avi.h b/libavformat/avi.h
index d1b5eb5..741781d 100644 (file)
--- a/libavformat/avi.h
+++ b/libavformat/avi.h
@@ -18,8 +18,7 @@ typedef struct CodecTag {
 void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const CodecTag *tags, int for_asf);
 int put_wav_header(ByteIOContext *pb, AVCodecContext *enc);
 int wav_codec_get_id(unsigned int tag, int bps);
-void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, 
-                    int has_extra_data);
+void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size); 
 
 extern const CodecTag codec_bmp_tags[];
 extern const CodecTag codec_wav_tags[];

diff --git a/libavformat/avidec.c b/libavformat/avidec.c
index ccf8f6e..6d23f07 100644 (file)
--- a/libavformat/avidec.c
+++ b/libavformat/avidec.c
@@ -187,7 +187,7 @@ static int avi_read_header(AVFormatContext *s, AVFormatParameters *ap)
 //                    url_fskip(pb, size - 5 * 4);
                     break;
                 case CODEC_TYPE_AUDIO:
-                    get_wav_header(pb, &st->codec, (size >= 18));
+                    get_wav_header(pb, &st->codec, size);
                     if (size%2) /* 2-aligned (fix for Stargate SG-1 - 3x18 - Shades of Grey.avi) */
                         url_fskip(pb, 1);
                     break;

diff --git a/libavformat/wav.c b/libavformat/wav.c
index 7fe2f8e..8130112 100644 (file)
--- a/libavformat/wav.c
+++ b/libavformat/wav.c
@@ -103,26 +103,44 @@ int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
     return hdrsize;
 }
 
-void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, 
-                    int has_extra_data)
+/* We could be given one of the three possible structures here:
+ * WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure
+ * is an expansion of the previous one with the fields added
+ * at the bottom. PCMWAVEFORMAT adds 'WORD wBitsPerSample' and
+ * WAVEFORMATEX adds 'WORD  cbSize' and basically makes itself
+ * an openended structure.
+ */
+void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size) 
 {
     int id;
 
     id = get_le16(pb);
+    codec->codec_id = wav_codec_get_id(id, codec->frame_bits);
     codec->codec_type = CODEC_TYPE_AUDIO;
     codec->codec_tag = id;
     codec->channels = get_le16(pb);
     codec->sample_rate = get_le32(pb);
     codec->bit_rate = get_le32(pb) * 8;
     codec->block_align = get_le16(pb);
-    codec->bits_per_sample = get_le16(pb); /* bits per sample */
-    codec->codec_id = wav_codec_get_id(id, codec->frame_bits);
-    if (has_extra_data) {
+    if (size == 14) {  /* We're dealing with plain vanilla WAVEFORMAT */
+        codec->bits_per_sample = 8;
+       return;
+    }
+    
+    codec->bits_per_sample = get_le16(pb);
+    if (size > 16) {  /* We're obviously dealing with WAVEFORMATEX */
        codec->extradata_size = get_le16(pb);
        if (codec->extradata_size > 0) {
+           if (codec->extradata_size > size - 18)
+               codec->extradata_size = size - 18;
             codec->extradata = av_mallocz(codec->extradata_size);
             get_buffer(pb, codec->extradata, codec->extradata_size);
-        }
+        } else
+           codec->extradata_size = 0;
+       
+       /* It is possible for the chunk to contain garbage at the end */
+       if (size - codec->extradata_size - 18 > 0)
+           url_fskip(pb, size - codec->extradata_size - 18);
     }
 }
 
@@ -259,7 +277,7 @@ static int wav_read_header(AVFormatContext *s,
     if (!st)
         return AVERROR_NOMEM;
 
-    get_wav_header(pb, &st->codec, (size >= 18));
+    get_wav_header(pb, &st->codec, size);
     
     size = find_tag(pb, MKTAG('d', 'a', 't', 'a'));
     if (size < 0)