Divide first audio buffer chunk into atomary bufffers.

This slightly simplifies decoder and removes potential
audio buffer overrun.

Originally committed as revision 7121 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 3 insertions, 19 deletions

diff --git a/libavcodec/vmdav.c b/libavcodec/vmdav.c
index 394d6d50d8..a9937144e4 100644
--- a/libavcodec/vmdav.c
+++ b/libavcodec/vmdav.c
@@ -519,12 +519,10 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
                                  uint8_t *buf, int buf_size)
 {
     VmdAudioContext *s = (VmdAudioContext *)avctx->priv_data;
-    unsigned int sound_flags;
     unsigned char *output_samples = (unsigned char *)data;
 
     /* point to the start of the encoded data */
     unsigned char *p = buf + 16;
-    unsigned char *p_end = buf + buf_size;
 
     if (buf_size < 16)
         return buf_size;
@@ -533,24 +531,10 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
         /* the chunk contains audio */
         *data_size = vmdaudio_loadsound(s, output_samples, p, 0);
     } else if (buf[6] == 2) {
-        /* the chunk contains audio and silence mixed together */
-        sound_flags = LE_32(p);
+        /* the chunk may contain audio */
         p += 4;
-
-        /* do something with extrabufs here? */
-
-        while (p < p_end) {
-            if (sound_flags & 0x01)
-                /* silence */
-                *data_size += vmdaudio_loadsound(s, output_samples, p, 1);
-            else {
-                /* audio */
-                *data_size += vmdaudio_loadsound(s, output_samples, p, 0);
-                p += s->block_align;
-            }
-            output_samples += (s->block_align * s->bits / 8);
-            sound_flags >>= 1;
-        }
+        *data_size = vmdaudio_loadsound(s, output_samples, p, (buf_size == 16));
+        output_samples += (s->block_align * s->bits / 8);
     } else if (buf[6] == 3) {
         /* silent chunk */
         *data_size = vmdaudio_loadsound(s, output_samples, p, 1);