Convert aac to wav to ??? to ac3

[tebasuna51]

@])r. Mario

Are you read this post completely?. The answer is there.
When you open this wavs in Soft Encode see the Time Line. Only 7:36, then Soft Encode finish Ok.
The problem is the bug in BeSplit with the field BlockAlign. Soft Encode see a 6 (must be 2) and read the wav three times short (20:50 / 3 = 7:36) and distorted.

With not huge wav6 files generated by Nero (o Foobar) is better split the file with Besweet:
BeSweet -core( -input g:\wav_6.wav -output g:\ -6CH )

Then you have: g:\FL.wav, C, FR, SL, SR, LFE
My Nero aac plugin put the channel order in the wav6: C, FL, FR, SL, SR, LFE
Then I must change FL.wav <-> C.wav
Foobar put the channel order in the wav6: FL, FR, C, LFE, SL, SR
The I must use the parameter for Besweet: -6ch (lowercase) and the names for the mono wav are ok.

[tebasuna51]

@Sakuya

1) Open output.wav in Hexedit and modify the Audioformat (second line, bytes 5 and 6) like this:

00|52 49 46 46 3c 00 5b 2f 57 41 56 45 66 6d 74 20|RIFF<.[/WAVEfmt
10|28 00 00 00 01 00 06 00 44 ac 00 00 30 13 08 00|(.......D...0...
20|0c 00 10 00 16 00 10 00 3f 00 00 00 01 00 00 00|........?.......
30|00 00 10 00 80 00 00 aa 00 38 9b 71 64 61 74 61|.........8.qdata
40|00 00 5b 2f

Save as output1.wav

2) Split the wav with Besweet (canonical header, no error in BLockAlign):
BeSweet -core( -input output1.wav -6ch )
Then you have: FL.wav, C, FR, SL, SR, LFE

3) Each mono wav must have 132415532 bytes and the header:

00|52 49 46 46 24 00 48 7e 57 41 56 45 66 6d 74 20|RIFF....WAVEfmt
10|10 00 00 00 01 00 01 00 44 ac 00 00 88 58 01 00|
20|02 00 10 00 64 61 74 61 00 00 48 7e

[])r. Mario]

@tebasuna51
i read the thread from the beggining, but must have not understood something, sorry

so are u saying that i should use besweet to make the 6 channel wave? or to split it?
if the bug is in besplit wont it still affect the wav?
yeah, i noticed right off that nreo wasnt assigning the channels correctly, i had to switch the FL with the C channel.
from what i understand your saying i should make the 6 channle wav from the aac file using besweet. ill try that
thx a lot for the help

[])r. Mario]

oops, just reread the post
u want me to split the 6 chan wav with besweet right?
sorry for the confusion, well, my confusion anyway

[Sakuya]

In BeSweet command line, I entered this but it won't work:

BeSweet -core( -D:\input.wav D:\output1.wav -6ch )

It only shows me definitions of each switch in BeSweet. I have a feeling I typed it wrong for the input and output.

[tebasuna51]

I know, my english is very bad and can't explain you correctly.

@])r. Mario
Three choices:

1) You can open directly the wav6 in Soft Encode. Then in the square icon at the left of the first track, click in the Center Channel. The second track is automatically set to Front Left channel. All is Ok. Encode.
(See PaulCat's comments in this post)

2) If you want to split the wav6 in 6 wav mono use BeSweet

3) If you want to split the wav6 in 6 wav mono with BeSplit, after you must correct the wav header field BlockAlign, like I say to Sakuya, in the six mono wav's.

@Sakuya

BeSweet -core( -D:\input.wav D:\output1.wav -6ch ) ???

If your modified wav, by Hexedit, is D:\output1.wav, use:

BeSweet -core( -input D:\output1.wav -output D:\ -6ch )

BeSweet puts automatically the file names to: D:\FL.wav, C, FR, SL, SR, LFE

[Sakuya]

Thanks Tebasuna, it worked! But however, my AC3 turned out to be shortened to 22 minutes and 59 seconds by Soft Encode. Why does it do that? Like Paulcat said, I set Soft Encode's Encode Settings to 48000Hz and UNCHECKED "Auto". Right now, here are the steps to get everything in perfect condition, including the sample rate:

1. AAC to 6ch WAV using FAAD2
2. Hexedit
3. 44100 to 48000 using SSRC
4. 6 mono WAV using BeSweet
5. AC3 using Soft Encode

Is there a way to compact my steps? For example, do the sample rate change in BeSweet instead of SSRC? That would make it much easier by using just one code in BeSweet to change the sample rate and splitting the 6ch WAV into 6 mono streams.

[tebasuna51]

I try with:

BeSweet v1.5b29 by DSPguru.
--------------------------
Using Shibatch.dll v0.24 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).

Logging start : 03/13/05 , 12:40:47.

BeSweet -core( -input g:\wav6_44.wav -6ch ) -shibatch( --rate 48000 )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : g:\wav6_44.wav
[00:00:00:000] | Output: FL, FR, SL, SR, C, LFE
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Source Sample-Rate: 44.1KHz
[00:00:00:000] | Dest. Sample-Rate: 48.0KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +---------------------
[00:00:20:000] Conversion Completed !
[00:00:05:000] <-- Transcoding Duration

Logging ends : 03/13/05 , 12:40:52.

But don't work, erroneous mono wav. Sorry.

[Sakuya]

Ok then, thanks for the help! I'll use the steps I have above then.

[Sakuya]

Okay, I gave up on the project that needed this sort of conversion several months ago. But now, I need it again. But just in time, Hexedit is not working on me!

I have a split MKV with AAC 5.1 sound. But I don't know if it is LC or HE. I used mkvtoolnix to merge them together and then used mkvextractGUI to get the AAC file. Then I used mp_rel to get the timecodes and the AVI. Fast forward to the 6ch WAV and I have to edit the header. Hexedit loads it and when the window comes up, it crashes. It's happened for the fourth time in a row. Is the WAV too big? It's 2.94GB. Is there another way? What determines that I need to change the header? Can I just skip the header part and proceed to changing the sampling rate to 48000 in SSRC?

[johnman]

I havent read the whole thread but i did see some problem with upsampling and the number of channels. If you got a VALLID wav, wavewizard can do upsampling and splitting streams to mono wav's in a single run. BTW wavewizard shows the number of channels in the wav if you wonder about this. Wavewizard can be found on http://forum.doom9.org/showthread.php?t=95265.

(im not trying to spam my program, but in this case i think it might help).

[Paulcat]

Originally Posted by Sakuya

Okay, I gave up on the project that needed this sort of conversion several months ago. But now, I need it again. But just in time, Hexedit is not working on me!

I have a split MKV with AAC 5.1 sound. But I don't know if it is LC or HE. I used mkvtoolnix to merge them together and then used mkvextractGUI to get the AAC file. Then I used mp_rel to get the timecodes and the AVI. Fast forward to the 6ch WAV and I have to edit the header. Hexedit loads it and when the window comes up, it crashes. It's happened for the fourth time in a row. Is the WAV too big? It's 2.94GB. Is there another way? What determines that I need to change the header? Can I just skip the header part and proceed to changing the sampling rate to 48000 in SSRC?

I have downloaded the latest STABLE versions of BeSweet and BeLight (a GUI for BeSweet), and it appears you can do a direct conversion from AAC to AC3 from there without any trouble. Have you tried this? It would be easier than making waves (heh heh).

[Sakuya]

I currently have BeSweet and BeSweet GUI. The GUI is v0.7b4. And last time I tried BeLight, the "Start" button keeps disappearing on me. I forgot how I got it to work again but every time I start the program, the button is at the very very very very bottom hidden by the Windows toolbar.

And are you sure the 6ch will be kept? In possibly good quality? So what must I do? I'm a bit hesitant on my BeSweet v1.4. What versions did you use? The latest stable BeLight is v0.21.

[johnman]

I would not use besweet ac3 encoding because it will heavily reduce the quality of the audio. But you can always try ofcourse.

[Sakuya]

Okay, I tried it out and it crashes every single time. Hexedit crashes too. Winhex won't let me save. Soft Encode won't let me open. This is the worst. Nothing works for me.

[johnman]

Originally Posted by Sakuya

Okay, I tried it out and it crashes every single time.

What crashes?

Originally Posted by Sakuya

Winhex won't let me save.

If a program has freezed an got a lock on the file you cant save or edit it. Sometimes you cant even open it. If you dont know how to shutdown processes you need to restart the computer. Doesnt winhex tell why you cant save?

[magicclue]

Hi there!

Sorry but this thread is way too long. So I will just try to tell two ways to transcode in an easy way with all channels in right order.

FAQ: Foobar2000 / Nero cannot transcode aac 5.1->wav correctly (hydrogenaudio). You'll end up with 8 multichannel wav.
Instead use aacDecDrop

Easy but long way without GraphEdit:

With aacDecDrop (drag and drop AAC/MP4 on it) to decode to a 6 multichannel wav.

Use BeSweet to separate into 6 mono Waves.
Attention: The naming of the files is wrong. You've to rename them to get correct channel mapping.

The channel mapping is as follows:

filename = channel name
FL.wav = Center
C.wav = Front Right
FR.wav = Front Left
SL.wav = Surround Right
SR.wav = LFE
LFE.wav = Surround Left


So the file C.wav is the Front Right channel!
e.g. rename c.wav to fr.wav

[magicclue]

Fast method with ffdshow:

get GraphEdit from: http://www.3ivx.com/download/windows.html look for the link to GraphEdit in the upper right corner.

get new FFDSHOW from: http://www.aziendeassociate.it/cd.asp?dir=/ffdshow download newest version of ffdshow

-install ffdshow
-activate decoding for aac!
-extract graphedit in a folder
-start graphedit
-drag&drop AAC-file in GraphEdit window
-click and remove "default direct sound device"
- click Menu graph->insert new filter->DirectShowFilter->Filewriter and "insert filter", choose filename "convert.ac3"
-right click on ffdshow->properties->output->AC3 and "allow direct-to-file output" and "don't use Waveformatextensible"
-click ok

[see figure 1]

-connect ffdshow filter with the file writer "convert.ac3" filter
-click play to start conversion and wait until its finished

[see figure 2]

If your output stutters it is very likely that your sampling rate is 44KHz and not 48KHz Some soundcard don't output 44KHz over S/PDIF correctly. To solve this, activate the resampler in ffdshow and set it to "Resample if sampling rate below... 48000Hz".

[johnman]

Originally Posted by magicclue

thx

[magicclue]

? Hey take it easy not personally. I'm stupid

for the guides.