Dealing with "raw" AAC and appsrc

[Brian Dilley]

MODS: sorry for the duplicate send, i used the wrong email address during
signup and had to change it.  Please disregard the last email :)

Hello friends.

I'm working on an RTMP ingestion server.  Currently I'm trying to figure
out how to handle the audio packets.  The audio packets come across as
AAC-HE 44kHz in an RTMP message (in this scenario).  They're packed in what
I think is called an FLV tag box?  Basically the first byte indicates the
codec (AAC in this case) and the second byte indicates whether or not it's
a configuration packet.  The configuration packets aren't very useful, and
i just ignore it for now (the very first audio packet is a configuration
packet).  So far so good - the rest of the RTMP message is the AAC data.
But it doesn't have an ADTS header, so I'm generating one (based on the
FFMPEG code).  So now i'm writing these packets (with a generated ADTS
header) into the following pipeline and was expecting it to dump decoded
audio into the file, but it results in error code 1: Internal data stream
error:

appsrc name=audioInput is-live=true caps=audio/mpeg ! faad ! filesink
name=output
location=/the/location.data


When I use the following pipeline the data makes it into the file, but it's
unplayable (mplayer just produces static):

appsrc name=audioInput is-live=true caps=audio/mpeg ! filesink name=output
location=/the/location.data


I've even tried both pipelines without the is-live and caps configuration
options on the appsrc.

I'm hoping someone out there has dealt with raw AAC and/or RTMP before and
knows what I'm overlooking.

Oh, and just incase it matters - I'm setting the presentation timestamp on
the Buffer before writing it to the AppSrc using the timestamp from the
RTMP client.


Thanks!
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