aac 5.1 conversion to anything(pulling hair out)

[dog-god]

i have mkv anime epsiode that has 5.1 aac audio
i been reading and experminting for the last 3 days but i cannot for the life of me find a way to convert this to 5.1 ac3 preferrable or mp3 downmix at worst

ive used mplayer - accdop - dbpoweramp - besweet - virtualdubmod - graphedit etc
if anybody has done this successfully ive appreciate some help
ive read the post concerning this on this forum but the sampling rate ,speed ,lenght vary using different methods
also there sems to be an issue with how the channels are mapped differently from ac3 5.1

if any want to play with the aac file(extracted from the mkv with MKVextractGUI_154) size 38mb link below


http://rapidshare.de/files/3564535/Track2.rar.html

grab file here

[E-Male]

ac3filter should do the job

[magicclue]

yep.
and there's a guide too.

See sticky in audio forum!!!

[SeeMoreDigital]

With the correct settings Foobar2000 can trancode 6Ch AAC-LC or HE streams (with .MP4 or .AAC file extentions) to say, 2Ch WAV, Vorbis, MP3, etc


Cheers

[magicclue]

Originally Posted by SeeMoreDigital

With the correct settings Foobar2000 can trancode 6Ch AAC-LC or HE streams (with .MP4 or .AAC file extentions) to say, 2Ch WAV, Vorbis, MP3, etc
Cheers

[Sorry but it cannot.
Foobar2000 does not decode AAC 5.1 correctly you'll end up with 8 channel wav! See Hydrogenaudio for more info on this...]


Use this guide: http://forum.doom9.org/showthread.ph...226#post666226

[SeeMoreDigital]

Originally Posted by magicclue

[Sorry but it cannot.
Foobar2000 does not decode AAC 5.1 correctly you'll end up with 8 channel wav! See Hydrogenaudio for more info on this...]

Use this guide or guide below that: http://forum.doom9.org/showthread.ph...226#post666226

You're right... I've just tried it and got an 8Ch WAV file, even after fiddling about with the DSP settings.... bummer!

However... all is not lost as I managed to use QuickTime Pro 7 to convert a RAW 6Ch AAC stream to an 2Ch WAV stream @ 44.1 KHz, which worked fine. I also tried converting to an 2Ch WAV stream @ 48.0 KHz, which did not work so well...


Cheers

[neo75903]

tried videolan? should do the job ...

[eb]

@dog-god,

I converted your sample to AC35.1 384kb/sec it is ca.70MB.
How to send it to you?
AS Doom9 wrote in news, and because the same reason I have no time to write more detailed answer.
Quality of this sample as ac3 is really nice.

eb

Edited: sorry it is AC3 2.0 44100

Again. Sample could be Ac3 5.1 484kb/s, 48000kbs samplerate, ca. 80MB

[Rockaria]

Originally Posted by neo75903

tried videolan? should do the job ...

I tried in a hope, but the best I got in the 'audio device' option was '2 front + 2 rear'.
The decoder in the vlc does not seem to be reading the format(HE+ AAC 5.1ch) correctly too.

[magicclue]

see post #5 Guide.
What else do you want?

[Rockaria]

Originally Posted by magicclue

see post #5 Guide.
What else do you want?

Hmm, I think it's for me.
Firstly, I wanted to leave a feedback to neo75903's message.
Also I like and enjoy the foobar2K dts/ac3->aac he+ in a step.

I have no doubt your guide must be very helpful to many and I appreciate that.
But among the many possible ways, I think I will choose the graphedit ffdshow audio(with possibly channel remappings) ac3 encoding-> file writer one.
What I wanted is the foobar aac decoder plugin updated soon.

[tebasuna51]

I know ac3filter/ffdshow can do the job.

But, what is the problem with Foobar?. I try with this Track2.aac and make a correct 6 channel wav (L,R,C,LFE,SL,SR - 44.1 KHz - 24m 20.21s). Whit DSP activated for a dpl downmix make a correct stereo wav.

I use Foobar2000 v0.8.3 (last stable). Maybe the problem aac 5.1 -> 8 channel wav is for v0.9 beta?

When I use aacDECdrop to decode this track2.aac it make a 6 channel wav (C,L,R,SL,SR,LFE - 22.05 KHz - 24m 20.21s). Incorrect channel order and downsampling to half frecuency.

When I use Faad (V2.1 beta) it inform about Track2.aac:
ADTS, 1460.256 sec, 215 kbps, 44100 Hz
And make a correct 6 channel wav (WAVE_FORMAT_EXTENSIBLE - L,R,C,LFE,SL,SR - 44.1 KHz - 24m 20.21s). Here the problem is: old soft (SoftEncode, ...) can't open WAVE_FORMAT_EXTENSIBLE.

[Rockaria]

I tested the foobar2k(the same v0.8.3) after reading this thread days ago. I made a 5.1ch he+aac(from a dts) and a 2.0 ch he+aac(from a mp3).
I opened the first one with the foobar2k and tried the 5.1ch he+aac encoding resulting in a turn off to the desktop. The second one transcoded without any problems.

As for the 'WAVE_FORMAT_EXTENSIBLE' format with the softEncode, the raw format read will do the trick if you don't care about the static noise in the beginning.
Or you can possibly read it as raw format with the first some blocks cut-off(manually).

[magicclue]

Originally Posted by tebasuna51

...But, what is the problem with Foobar?. I try with this Track2.aac and make a correct 6 channel wav (L,R,C,LFE,SL,SR - 44.1 KHz - 24m 20.21s). Whit DSP activated for a dpl downmix make a correct stereo wav...

I use Foobar2000 v0.8.3 (last stable). Maybe the problem aac 5.1 -> 8 channel wav is for v0.9 beta?
...
When I use aacDECdrop to decode this track2.aac it make a 6 channel wav (C,L,R,SL,SR,LFE - 22.05 KHz - 24m 20.21s). Incorrect channel order and downsampling to half frecuency...

Ahh.. guess what:

You downmixed and lost all discrete channel infos.
But the question was AAC 5.1 -> AC3 5.1
So there's no downmix.

If you decode AAC 5.1 with Foobar2000 0.8.3 to multichannel WAV you'll end up with a 8ch WAV. That's the problem.
Might work if you downmix to Dolby Pro Logic. But then you'll only need 2 channels.

For aacDecDrop and channel mapping see my guides posted above.
Or my ffdshow/graphedit guide (also above)

[Rockaria]

Yeah, I found now you also included the graphedit with ffdshow audio/ac3filter method.
Having many DSP plugins, the ffdshow audio dsfilter seems to be a better option.

The 2ch (downmix) AAC decoding+encoding surely works in the foobar2k. Also I've noticed the HE+AAC 5.1ch play in the foobar is not stable(somewhat echoing : maybe the 8ch effect).

The foo_faac.dll(192,000 byte) is dated 5/16/2004.
(I could find nothing like foo_faad.dll)

[Rockaria]

I did some more test on the foobar2k AAC(Mp4) decoding(is said included in the standard input).
The channel order is messed up unlike other decoder(ac3filter or ffdshow audio plays correctly).
But I was not sure whether it decodes to 8ch or not, the mixer spcturum showed 6ch bars moving.
The transcoding failed as I said before.(I suspect 8ch)

When I used the nero decoder(added ';aac;mp4' to the decoder extention), the channel order messed up differntly, but finished the encoding(I suspect 6ch).
I guess the nero decoder output can be used for transcoding if we can arrange the channel order anyhow.

[tebasuna51]

Originally Posted by magicclue

You downmixed and lost all discrete channel infos.
But the question was AAC 5.1 -> AC3 5.1
So there's no downmix.

If you decode AAC 5.1 with Foobar2000 0.8.3 to multichannel WAV you'll end up with a 8ch WAV. That's the problem.

One more time.
"I try with this Track2.aac and make a correct 6 channel wav (L,R,C,LFE,SL,SR - 44.1 KHz - 24m 20.21s). "
End point.

I make, also, a 6 channel wav with Faad and other with GraphEdit-ffdshow. This two 6 channel wav are identical 100% (same decoder).

When compare wav6_foobar with wav6_ffdshow aren't identical but with inaudible differences. The most notables are:
a 50 ms. 22.05 KHz -27 dB
a 50 ms. 22.05 KHz -76 dB
the rest are under -90 dB

Then my question is: When fail the foo_aac_decoder?
All my samples in aac 5.1 are well decoded with Foobar.

Originally Posted by Rockaria

I opened the first one with the foobar2k and tried the 5.1ch he+aac encoding resulting in a turn off to the desktop.

I'm interested in Foobar aac decoder.

Originally Posted by Rockaria

As for the 'WAVE_FORMAT_EXTENSIBLE' format with the softEncode, the raw format read will do the trick if you don't care about the static noise in the beginning.
Or you can possibly read it as raw format with the first some blocks cut-off(manually).

If you change, with WinHex, the field wav header AudioFormat 0xFFFE ('WAVE_FORMAT_EXTENSIBLE', at Offset 0x20) with the SubType (Offset 0x44) 0x001 (Integer PCM) or 0x003 (Float 32bit) the resulting wav file can be opened for SoftEncode without raw or noise (the header misunderstand) problems.

[Rockaria]

Originally Posted by tebasuna51

I'm interested in Foobar aac decoder..

I admire your cutting & editing ability.
What is wrong with my method aac-encoding -> aac-decoding+aac-encoding to check the foobar aac-decoding for the transcoding?

Originally Posted by tebasuna51

Here the problem is: old soft (SoftEncode, ...) can't open WAVE_FORMAT_EXTENSIBLE.

I also admire your fresh memory. I just told you a clear method(in my memory).
Then what was your problem exactly?

Originally Posted by tebasuna51

Then my question is: When fail the foo_aac_decoder?

Again, foobar aac decoder has no problem in decoding(playing) except the channel mess-up. When the decoder is used with the transcoding together(to any format), it errored out showing the debug screen, I suspect it is because of the extra two channels, as is said.

[dog-god]

i havent gone away - been reading what my betters have been posting here with interest
magicclue that graphedit tutorial is nice but there is one prob
it dosent ouput 48k sample rate only what has been input
gspot reports 44000Hz CBR 448 kb/s total (5 chnls)
and dvd mastero refuses to take the file as the sample rate is wrong

one last this is about the channel mapping - when i do it all the voice is coming from my left front speaker ?
when i leave changing the channel mapping - it seems move like what i should have

when someone encodes to acc is it possibel to leave the channel mapping the same as ac3, could that have been done here

[Rockaria]

OK, my first contribution to dog-god, the original poster.
The ffdshow audio ds filter has many DSPs to use for the ac3 encoding.
Among them, the resample option can be used to make the 48K ac3 output.
I set the 'resample' in the 'Audio Settings' is cheked set as 'resample to 48000' 'when the source is lower than 48000'.