[FFmpeg-devel] help --- RTP AAC packetizer problem

[Manas Bhattacharya]

Hi
  i was trying to stream AAC audio to rtp from the ffmpeg version 11870  but
neither Quicktime nor ffplay was able to play the stream .
 the rtp aac demuxer is the latest.
         I  debugged through ffplay and i found out the au_header_size is
taking the value of 0  in rtp_parse_mp4_au
call from rtp_parse_packet in rtpdec.c and the function is  returning
without completion.  In fact the p[0] value in rtp_aac.c is taking the value
0 which is the access unit header size. Mu question is why is the header
size 0 ? and is it the only problem of any player not being able to play the
streams. I am attaching the ffmpeg command the trace  . Can any one  help ??

  ffmpeg -i sample.mpeg  -acodec libfaac -vn -f rtp rtp://localhost:8090

ffmpeg  -re -i sample.mpeg -acodec libfaac -ac 2 -vn -f rtp
rtp://localhost:8090
FFmpeg version SVN-r11870, Copyright (c) 2000-2008 Fabrice Bellard, et al.
  configuration: --enable-libfaac --enable-pthreads --enable-memalign_hack
  libavutil version: 49.6.0
  libavcodec version: 51.50.0
  libavformat version: 52.7.0
  libavdevice version: 52.0.0
  built on Feb 28 2008 18:38:48, gcc: 3.4.4 (cygming special, gdc 0.12,
using dm
d 0.125)
Input #0, mpeg, from 'sample.mpeg':
  Duration: 00:03:04.9, start: 0.661544, bitrate: 703 kb/s
    Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 320x240 [PAR 200:219 DAR
800
:657], 500 kb/s, 29.97 tb(r)
    Stream #0.1[0x1c0]: Audio: mp2, 44100 Hz, stereo, 192 kb/s
Output #0, rtp, to 'rtp://localhost:8090':
    Stream #0.0: Audio: libfaac, 44100 Hz, stereo, 64 kb/s
Stream mapping:
  Stream #0.1 -> #0.0
[libfaac @ 0x75a530]AAC with no global headers is currently not supported
SDP:
v=0
o=- 0 0 IN IPV4 127.0.0.1
t=0 0
s=No Name
a=tool:libavformat
c=IN IP4 localhost
m=audio 8090 RTP/AVP 96
size=     607kB time=78.7 bitrate=  63.2kbits/s