Fix two checks in the AAC packetiser, which were too conservative

Patch by Martin Storsjö (martin AT martin DOT st)

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@18400 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

diff --git a/libavformat/rtp_aac.c b/libavformat/rtp_aac.c
index bfff56a105d3ec976dc8b7ca594b778389924478..1a9265b1de2e4c9595cb3b2e596f2d6f31ff74a3 100644 (file)
--- a/libavformat/rtp_aac.c
+++ b/libavformat/rtp_aac.c
@@ -39,7 +39,7 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
 
     /* test if the packet must be sent */
     len = (s->buf_ptr - s->buf);
-    if ((s->num_frames == MAX_FRAMES_PER_PACKET) || (len && (len + size) > max_packet_size)) {
+    if ((s->num_frames == MAX_FRAMES_PER_PACKET) || (len && (len + size) > s->max_payload_size)) {
         int au_size = s->num_frames * 2;
 
         p = s->buf + MAX_AU_HEADERS_SIZE - au_size - 2;
@@ -59,7 +59,7 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
         s->timestamp = s->cur_timestamp;
     }
 
-    if (size < max_packet_size) {
+    if (size <= max_packet_size) {
         p = s->buf + s->num_frames++ * 2 + 2;
         *p++ = size >> 5;
         *p = (size & 0x1F) << 3;