Audio encode bitrate

Hello,

I have a custom DM8148 board, In which I encode audio.

Firstly I capture audio (wav file)  arecord -f dat -d 10 audio_encode.wav

Then I use the audio encode example from omx (without any modifications)

audio_encode  -i audio_encode.wav -o audio_encode.m4a  -c aaclc -n 2 -b 256000 -s 48000 -f ADTS

Following is the log:

input_file: audio_encode.wav
output_file: audio_encode.m4a
codec: aaclc
no of channnels :2
bitrate : 256000
samplerate : 48000
outputformat :ADTS
===============================
Audio Encoder Example
Component transitioned from Idle to Execute
FileSize : 1920044
etb called
encoded: 000468
Received End of stream
Going to Idle
Going to Loaded
Freeing buffers
Component transitioned from Idle to Loaded
Audio Encode Test End
Done!

Now when I check the bitrate its variable:

prabhakar@tango-charlie:/media/prabhakar/646F-8F52$ mediainfo audio_encode.m4a
General
Complete name                            : audio_encode.m4a
Format                                   : ADTS
Format/Info                              : Audio Data Transport Stream
File size                                : 313 KiB
Overall bit rate mode                    : Variable

Audio
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format version                           : Version 4
Format profile                           : LC
Bit rate mode                            : Variable
Channel(s)                               : 2 channels
Channel positions                        : Front: L R
Sampling rate                            : 48.0 KHz
Compression mode                         : Lossy
Stream size                              : 313 KiB (100%)

When I right click and see the properties and see Bitrate: N/A

Because of this some of the swf players are not able to play this audio.

Is there something in encoder which can force the bitrate to be fixed and not variable ?

audio_encode.m4a

Cheers,

--Prabhakar Lad