Problems with an AAC filie

[Xayd]

Hi, I've got a HDTV capture that has AAC audio, and I'm trying to convert it to ac3 since my receiver doesn't decode AAC.

I suspect there are problems with this stream but I don't know which is causing the issue(s) I'm having or how to go about fixing them.

1) the stream header is wrong, players detect it as stereo but it's actually 5.1. i'm not sure if this causes a decoding problem or not.

2) using the two filters that ffdshow has for decoding AAC, along with graphedit to transcode the output to ac3, I get one of two errors. with libfaad2 it halts at 00:24:30, which doesn't have any audible error when listening to the file, with realaac it halts at 00:3:21, which also doesn't have any audible error when listening to the file. the total length of the aac file is just under two hours.

3) i tried repairing the file with besplit, which identified 3 errors in the stream, but attempting the same conversion on the output with graphedit and ffdshow results in the same problem as before, either a halt at 3:21 or 24:30 depending on which decoder is used.

here's the log from the besplit repair of the file...


BeSplit v0.82 by DSPguru.
--------------------------
Logging start : 12/10/05 , 19:15:13.

C:\Program Files\BeSweet\BeSplit.exe -core( -logfile F:\HDTV\clapton-324ms.LOG -input F:\HDTV\clapton-324ms.aac -prefix F:\HDTV\clapton-324msFIX -type aac -a -fix )

[00:00:00:000] +------- BeSplit -----
[00:00:00:000] | Input : F:\HDTV\clapton-324ms.aac
[00:00:00:000] | Output Prefix : F:\HDTV\clapton-324msFIX
[00:00:00:000] +---------------------
[00:03:21:536] Stream error : Sync found after 563 bytes
[01:09:15:882] Stream error : Sync found after 508 bytes
[01:09:17:141] Stream error : Sync found after 251 bytes
[01:59:52:533] | Writing F:\HDTV\clapton-324msFIX01.aac
[01:59:52:533] +---------------------
[01:59:52:533] Operation Completed !
[00:00:56:000] <-- Process Duration
Logging ends : 12/10/05 , 19:16:09.

any ideas?

[bond]

how did you capture the file and how did you extract the .aac out of it?

[Xayd]

the file is a transport stream (with mpeg2 video), i demuxed the AAC file with dgindex set to "demux all tracks".

[bond]

the file is a transport stream (with mpeg2 video), i demuxed the AAC file with dgindex set to "demux all tracks".well i never tried dgindex, but maybe it doesnt do it 100% correctly?

also try ffmpeg and mplayer for demuxing

[Xayd]

mplayer gets the same error at the same point that ffdshow does with the error "gain control not yet implemented".

[bond]

mplayer gets the same error at the same point that ffdshow does with the error "gain control not yet implemented".thats the typical faad2 error if something isnt right with the stream

try ffmpeg and mplayer _for demuxing_

[WarrenS]

I thought it an amazing coincidence that the AAC file I'm having a problem with also stopped at 3:21 when I tried to decode to .wav, and then I noticed Xayd and I must both be dealing with the same show.

The show is an HDTV capture of Clapton Live in Japan, and I too used DGIndex to demux because nothing else would accept this program stream. VideoReDo, which I usually use to join .ts files into one program wouldn't load the files, and nothing else would either.

I have tried ffmpeg, faad, dbPowerAMP, any number of those "all in one we convert anything to anything" programs, and possibly some things I can't recall right now, since it's been a 3 day project so far.

I get the same "gain control not yet implemented" error in faad when it starts to decode. Every other program cuts the output file at 3:21, when the show was two hours long. WinAmp plays the AAC rip fine, no problems, all the way through.

I hope this gives someone an insight into the problem, 'cause I'm stumped.

Edit:
I do believe I have found an answer to the problem: It's really just a work-around, but since the audio played fine in WinAmp, I loaded up one of those audio recorder apps (I used River Past Audio Converter) (Google "record from soundcard") and played the entire sound file in WinAmp while recording the output as a wav file.

From here, I hope to encode to AC3 as normal and mux the sound back in before burning to DVD.