Cannot convert ADTS AAC file to anything

[Killer3737]

I have the following audio stream taken from a video:

Format : ADTS
Format/Info : Audio Data Transport Stream
File size : 112 MiB

Audio
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : SSR
Bit rate mode : Variable
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 KHz
Compression mode : Lossy
Stream size : 112 MiB (100%)

Which I'm trying to decode to WAV, AC3, anything. I've normally no problem with AAC but this seems to be an ADTS file and no matter what program I try, nothing. VLC won't even play it when it's on it's own, it says "No suitable decoder module: VLC does not support the audio or video format "mp4a". Unfortunately there is no way for you to fix this." It plays it when it's still muxed with the video though.


So, any ideas for getting this converted? I've googled and tried every program I can find but no luck at all.

[Killer3737]

Just for future reference in case anyone else looks for this, I finally got it converted with MeGUI.

[El Heggunte]

Just for the record:

Audio

Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : SSR

en.wikipedia.org/wiki/MPEG-4_Part_3#AAC-SSR says:

AAC Scalable Sample Rate was introduced by Sony to the MPEG-2 Part 7 and MPEG-4 Part 3 standards. It was first published in ISO/IEC 13818-7, Part 7: Advanced Audio Coding (AAC) in 1997. The audio signal is first split into 4 bands using a 4 band polyphase quadrature filter bank. Then these 4 bands are further split using MDCTs with a size k of 32 or 256 samples. This is similar to normal AAC LC which uses MDCTs with a size k of 128 or 1024 directly on the audio signal.

The advantage of this technique is that short block switching can be done separately for every PQF band. So high frequencies can be encoded using a short block to enhance temporal resolution, low frequencies can be still encoded with high spectral resolution. However, due to aliasing between the 4 PQF bands coding efficiencies around (1,2,3) * fs/8 is worse than normal MPEG-4 AAC LC.

MPEG-4 AAC-SSR is very similar to ATRAC and ATRAC-3.

Why AAC-SSR was introduced

The idea behind AAC-SSR was not only the advantage listed above, but also the possibility of reducing the data rate by removing 1, 2 or 3 of the upper PQF bands. A very simple bitstream splitter can remove these bands and thus reduce the bitrate and sample rate.

Example:
4 subbands: bitrate = 128 kbit/s, sample rate = 48 kHz, f_lowpass = 20 kHz
3 subbands: bitrate ~ 120 kbit/s, sample rate = 48 kHz, f_lowpass = 18 kHz
2 subbands: bitrate ~ 100 kbit/s, sample rate = 24 kHz, f_lowpass = 12 kHz
1 subband: bitrate ~ 65 kbit/s, sample rate = 12 kHz, f_lowpass = 6 kHz

Note: although possible, the resulting quality is much worse than typical for this bitrate. So for normal 64 kbit/s AAC LC a bandwidth of 14–16 kHz is achieved by using intensity stereo and reduced NMRs. This degrades audible quality less than transmitting 6 kHz bandwidth with perfect quality.