Convert aac to wav to ??? to ac3

[blizzard]

Ok guys.Thank you very much for the help!

[Aragorn Skywalker]

Hi,

I'm trying to convert a 5.1 AAC into a 5.1 ac3 file for use in DVDLab. I'm having trouble following the guides. I'm new at all of this so any help would be great.

I have FFDSHOW and Graphedit installed.

When I drag and dog the AAC file into Graphedit it gives me this error:

Could not construct a graph for the file
-Have you installed all needed filters?
-Note that the 'render file' menu option cannot read *.grf files

Classfactory cannot supply requested class (return code 0x80040111)

What am I doing wrong?

[Livesms]

So is there any correct way to get best results for aac->ac3 conversion

How can I get right 6 wav mono files and no to mess with channels.
I will use Sony Encode to convert 6 mono wav to AC3 5.1

[Livesms]

I have 6channel wav file (from aacDECdrop)

When I try smth like this

D:\1>D:\1\BeSweet.exe -core( -input D:\1\1.wav -output D:\1\ -6ch )
BeSweet v1.4 by DSPguru.
--------------------------

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : D:\1\1.wav
[00:00:00:000] | Output: FL, FR, SL, SR, C, LFE
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:00:00:032] Conversion Completed !
[00:00:00:000] <-- Transcoding Duration


Visit DSPguru's Homepage at :
http://DSPguru.doom9.net/

And after that there is 6 files all 3116 bytes
3*116 C.wav
3116 FL.wav
3116 FR.wav
3116 LFE.wav
3116 SL.wav
3116 SR.wav

Why? Is therу any way to fix this problem and get right 6 mono wav files.

[tebasuna51]

So is there any correct way to get best results for aac->ac3 conversion

How can I get right 6 wav mono files and no to mess with channels.
I will use Sony Encode to convert 6 mono wav to AC3 5.1
Then you need a decoder and split the wav6chan in mono Wav's.

Best is a forbidden word in this forum then I propose you three decoder and two split methods, there are more of course.

Decoders:
- Foobar2000 (http://www.foobar2000.org/), the output can be piped to the spliter.
- Faad2 (http://www.rarewares.org/aac.html), the output can't be piped then you need a intermediate wav6chan (can be >4GB).
- your default DirectShow aac decoder (properly configured), the output can be piped to the spliter.

To split in mono wav's:
- WaveWizard (http://www.rarewares.org/wavewiz/wavewizardv0.54b.zip), accept any kind of wav6chan (int16-24-32, float32-64, WAVE_FORMAT_EXTENSIBLE, >4GB) like input.
- Wav2mono, written for Bepipe/BeHappy but can be used also with Foobar2000 or in command line mode with support also for int16-24-32, float32-64, WAVE_FORMAT_EXTENSIBLE, >4GB.

To use wav2mono with Foobar2000 see my post (http://forum.doom9.org/showthread.php?p=901020#post901020)
The "3) convert a .dts or .ac3 to 6 mono wav" is valid for any number of channels and any format (like aac) supported by Foobar. In this post you have also a link to obtain wav2mono.

To use wav2mono with DirectShow aac decoder you need Bepipe/Behappy, AviSynth and .NET Framework v2.0 and:
bepipe --script "DirectShowSource(^yourpath\your.aac^)" | wav2mono - prefix.wav -ignorelength

To use wav2mono in command line mode:
faad -o intermediate.wav input.aac
wav2mono intermediate.wav -ignorelength

[Livesms]

To use wav2mono with DirectShow aac decoder you need Bepipe/Behappy, AviSynth and .NET Framework v2.0 and:
bepipe --script "DirectShowSource(^yourpath\your.aac^)" | wav2mono - prefix.wav -ignorelength

To use wav2mono in command line mode:
faad -o intermediate.wav input.aac
wav2mono intermediate.wav -ignorelength

I used aacDECdrop.exe to convert aac track to 6ch wav track (4.34gb size)
And then
besplit -core( -input 1.wav -prefix mono -demux -type wav )

and mono01.wav - mono06.wav was prodused.

Then
Open "D:\1\mono01.wav" in-place (your path and name)
Goto 0x14 (Offset of Audiformat)
Write 0x0100 (PCM, lower byte first)
Save
Exit


But tebasuna51 tells (http://forum.doom9.org/showthread.php?p=616235#post616235) that order is FL, FR,C,LFE,SL,SR - but i my case center (with speach ) is mono01.wav and LFE (moslty silent) is mono06.wav

What shall I do
Can I assume
mono01.wav - C
mono02.wav - FL
mono03.wav - FR
mono04.wav - SL
mono05.wav - SR
mono06.wav - LFE
Is there any way to found out this?

PS: Can you give me a link to wav2mono. Cause tebasuna51 link posted here (http://forum.doom9.org/showthread.php?p=886080#post886080) is temp. down :(

[tebasuna51]

I have 6channel wav file (from aacDECdrop)
aacDECdrop produce a wrong channelmapped wav (C,FL,FR,SL,SR,LFE), must be FL,FR,C,LFE,SL,SR. Also the samplerate is half than original for low bitrate aac (for instance with aac-HE)
When I try smth like this

BeSweet.exe -core( -input D:\1\1.wav -output D:\1\ -6ch )
BeSweet don't support WAVE_FORMAT_EXTENSIBLE and int wav >2GB

[Livesms]

aacDECdrop produce a wrong channelmapped wav (C,FL,FR,SL,SR,LFE), must be FL,FR,C,LFE,SL,SR. Also the samplerate is half than original for low bitrate aac (for instance with aac-HE)

BeSweet don't support WAVE_FORMAT_EXTENSIBLE and int wav >2GB

aacDECdrop prodused me a 6ch wav file, which I splited to 6 mono with besplit -core( -input 1.wav -prefix mono -demux -type wav )
Then Winhex script was applied.
If I will open it with Sonic Foundry Soft Encode in order of
"mono02.wav" "mono03.wav" "mono01.wav" "mono06.wav" "mono04.wav" "mono05.wav"
Will it be ok? Is it the same result as
faad -o intermediate.wav input.aac
wav2mono intermediate.wav -ignorelength

And what about wav2mono?
Where can I downloader wav2mono - Mytempdir removed file.

[tebasuna51]

The AudioFormat in mono wav's produced by BeSplit I think is correct because the wav6 is not WAVE_FORMAT_EXTENSIBLE.
Check the BlockAlign at 0x20, must be 2 (16 int mono) instead 12.

Your assumed channels are correct.

A temp link to WavUtil_2.zip (http://www.mytempdir.com/1128809)

[tebasuna51]

The http://rapidshare.com/files/3572747/WavUtil.zip.html in the mentioned post still work for me.

Sonic Foundry SoftEncode default order is FL,C,FR,SL,SR,LFE then your monowav files must be 02, 01, 03, 04, 05, 06

[Moti172]

Hi
I tried to follow the instructions above:

Extract the 6ch AAC from MKV with MKVExtractGUI.
Use:
faad -o intermediate.wav input.aac
wav2mono intermediate.wav -ignorelength

I got 6 wav files at the exact same size.
When I tried to play them, it seems that the one with the name BL contain the speech, and the others had low music.

My questions are:
1. Does wav2mono separate the channels in the same order as the files name?
2. Is it possiable that the channels in the AAC file are incorrect? if so, is there away to know what is the most likely possible mistake?
3. Is there any free software to encode the 6ch into AC3?

Thanks

[tebasuna51]

1. Does wav2mono separate the channels in the same order as the files name?
Nope, the order is:
Channel 1 -> FL (Front Left)
Channel 2 -> FR (Front Right)
Channel 3 -> FC (Front Center)
Channel 4 -> LF (Low Frequency)
Channel 5 -> BL (Back Left)
Channel 6 -> BR (Back Right)

2. Is it possible that the channels in the AAC file are incorrect? if so, is there away to know what is the most likely possible mistake?
If the dialogs (Channel 3 - FC) appear in Channel 5 - BL after Faad (output: MS defaults defined in WAVE_FORMAT_EXTENSIBLE) and Wav2mono the only explanation is the original AAC was wrong mapped.
The ac3 order is (FL-FC-FR-BL-BR-LF) and the internal aac (FC-FL-FR-BL-BR-LF), as you can see the first 3 channels (fronts) can be mistaken but never go to channels 4-5-6 (backs and LF). The error must be a manual mistake.

3. Is there any free software to encode the 6ch into AC3?
Aften is the free encoder to use, but need a unique 6chan wav.
You can use WaveWizard to merge the 6 mono wav's.

[Moti172]

Thank you for your answers.
It seems that the AAC's channels order was wrong.
:thanks:

[naboth]

Hello,

First I just want to thank everybody for contributing to this thread, it really helped me with a project I was very excited about. But I've come across a problem with a particular 5.1 aac file, and was wondering if anyone could help.

Using faad as outlined above, I've read that if it gives you an error message, you can know that there is a error with the aac file that faad can't process. Well, I am getting an error message, and I really need to transcode this aac file to ac3, so hopefully somebody can help me with what I need to do to get around this issue.

faad stops decoding around 15% and spits out the error message: "Error: Maximum number of scalefactor bands exceeded."

So is there any way to get around this? Any fix, or another app for decoding the aac to a wav file? TIA!

[tebasuna51]

You can try with Foobar2000 or any DirectShow decoder (coreaac, ffdshow, ...)

[ImAhNoBoDy]

I'm trying to do the graphedit way in this guide, but can't get to work. I was able to output to a 6 channel wav file through graphedit but not a 6 channel ac3. I have ffdshow audio configs in the output as follows:

http://img.photobucket.com/albums/v395/SeX1eStAsaBa/Others/ffdshowoutput.jpg

I'm not sure if this is an issue too, but when I check ac3 output ffdshow doesn't show up when I play a wav file. When I have ac3 output turned on, I drag the wav file in graphedit and ffdshow doesn't appear. If I uncheck output for ac3 of course wav and ac3 files are playable. Is this normal? I've tried all kinds of ways but can't get it to work. I know I can get a 6 channel ac3 with virtualdub but I wanted to try it with graphedit.


My graph is as follows:

blah.wav --> wav parser --> ffdshow audio decoder (with ac3 output on)--> filewriter

[tebasuna51]

I'm trying to do the graphedit way in this guide, but can't get to work. I was able to output to a 6 channel wav file through graphedit but not a 6 channel ac3.
Work for me. Maybe we have different ffdshow version or DS configuration.
If you want encode to ac3 file (not for SPDIF output) don't use ffdshow like encoder (always -6dB low volume) use Aften instead.

[ImAhNoBoDy]

^darn, we might have different version then. I have ffdshow_rev1734_20071229_xxl. Which version of ffdshow do you have?

I'm pretty sure ac3filter should output to ac3 using graphedit right?

[tebasuna51]

ffdshow_beta3_rev1324_20070701_clsid

When I check ac3 output, the 16,24, ...bit integer go unchecked (I can't obtain your image)

I don't use Ac3Filter

[ImAhNoBoDy]

I use xxl's build. I'll see if clsid build works or not. I tried to turn off 16, 24, etc but it seems like one of them NEEDS to be checked. If it works I'll post what I did.

[nautilus7]

May i ask why do you both use such an old version of ffdshow?

[ImAhNoBoDy]

Nope, no go. I tried ffdshow_rev1771_20080113_clsid, but whenever I would play a 6 channel wav file it would crash on windows media player...same with graphedit. I don't think this version is capable to play 6 channels cause in the mixer the most is 5 channels showing.

I tried ffdshow_rev1771_20080113_clsid again to see if when pulling up directshow filters in graphedit, it would crash. It did so I tried to version before that which was ffdshow_rev1732_20071228_clsid_sse_icl10 and still unable to get ac3 output. This time 16, 24, etc bits are not on and ffdshow still will not output to ac3.

I guess I'll just stick with virtualdub for now or until I can figure something out.

[tebasuna51]

I don't think this version is capable to play 6 channels cause in the mixer the most is 5 channels showing.

Really strange.

I guess I'll just stick with virtualdub for now or until I can figure something out.

I don't know how you can use VirtualDub to encode wav -> ac3.

[ImAhNoBoDy]

Well the audio stream has to be shorter than a random video stream. I'll get like a 2 hour movie, then get a wav source from the audio menu. I then compress it using ac3 acm. Save as wav, and I have a ac3 file. In VirtualdubMod it doesn't work cause it keeps on saying that it can't decompress the audio stream.

Yea I thought it was also strange that you were able to decode the 6 channel audio files with the same version of ffdshow too. Are you also using Haali? Cause I'm not.

[Robertus]

sorry guys, i have a problem

i try with graphedit but i have this error:

these filter cannot agree on a connection, verify type compatibility of input pin and output pin.

this message, when i try to connect .aac output with ffdshow audio decoder input.

can you help me?

i use the last ffdshow and i set it:
Codecs:

AC3: liba52
AAC: realaac

OUTPUT:

AC3 (SPDIF encode mode) enable

bitrate 448

coonect to any filter
dont use waveform.... enable
allow direct-to-file output enable

please help me :(

[tebasuna51]

The .aac output is from an .aac file or a splitter?

If is from a file maybe you need an AAC parser.

The target is convert to an .ac3 file or play through SPDIF?

If you want a conversion I suggest you use Foobar2000 or an AviSynth method (BeHappy/SoundOut/Wavi) instead ffdshow.

[Robertus]

The .aac output is from an .aac file or a splitter?

If is from a file maybe you need an AAC parser.

The target is convert to an .ac3 file or play through SPDIF?

If you want a conversion I suggest you use Foobar2000 or an AviSynth method (BeHappy/SoundOut/Wavi) instead ffdshow.

foobar don't convert in .ac3

another program? or method avysinth?

[Robertus]

i use nero to convert .aac in .ac3, but i don't know if this .ac3 is fine about quality.

[digifruitella]

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".

I appreciate your write up, but it's very confusing...

1. "choose filename "convert.ac3""

huh? I get a box popping up getting me to open something, yeah, but where do I FIND this "convert.ac3" file

2. "[see figure 1]

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

Absolutely, doesn't make sense, because figure one doesn't show how to do that..

[tebasuna51]

I appreciate your write up, but it's very confusing...

1. "choose filename "convert.ac3""

huh? I get a box popping up getting me to open something, yeah, but where do I FIND this "convert.ac3" file

If you insist in this method:

FileWriter ask you the name of output file, you don't need find nothing only select the name you want.

2. "[see figure 1]

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

Absolutely, doesn't make sense, because figure one doesn't show how to do that..

- Conect filters in GraphEdit is drag (left mouse click and drag) the output pin from one filter to the input pin of the other.

[donnyboy]

i've done exactly as dictated above and my graphedt keeps crashing no matter what. i recently bought a creative DDTS-100 decoder and have configured AC3/DTS output over SPDIF with AC3 encode mode enabled using the latest FFDShow. my soundcard is a Creative SB Live 24Bit. Can someone tell me whats wrong? Any help will be most appreciated. Thank You:)

[tebasuna51]

i've done exactly as dictated above and my graphedt keeps crashing no matter what. i recently bought a creative DDTS-100 decoder and have configured AC3/DTS output over SPDIF with AC3 encode mode enabled using the latest FFDShow. my soundcard is a Creative SB Live 24Bit. Can someone tell me whats wrong? Any help will be most appreciated. Thank You:)

This thread is old and a little confuse. Initially was an aac -> ac3 transcode method.

Can you explain your problem? (maybe open a new thread)
Because I don't know for what you need a DDTS-100 decoder and a soundcard Creative SB Live 24Bit.

[donnyboy]

This thread is old and a little confuse. Initially was an aac -> ac3 transcode method.

Can you explain your problem? (maybe open a new thread)
Because I don't know for what you need a DDTS-100 decoder and a soundcard Creative SB Live 24Bit.

Well i bought a DDTS-100 for my PS3 and decided to hook it up to my pc when watching 1080p HD movies. You see my pc is a measly 3.0GHz Prescott P4 with 512Mb of RAM and 1080p HD movies really gives it a run for its money even with the CoreAVC codec properly installed and configured. So i decided to stream AC3 through S/PDIF to the DDTS-100 decoder and save CPU usage. That was the whole point behind hooking up my pc and the decoder.

I wanted to convert 6 channel LC AAC to its equivalent AC3 format and whenever i follow the above set of instructions Graphedt always crashes and locks up. Is there anyway to solve this. I have the Dec'07(rev1723) version of FFDShow installed.

[tebasuna51]

...So i decided to stream AC3 through S/PDIF to the DDTS-100 decoder and save CPU usage. That was the whole point behind hooking up my pc and the decoder.

I wanted to convert 6 channel LC AAC to its equivalent AC3 format and whenever i follow the above set of instructions Graphedt always crashes and locks up. Is there anyway to solve this. I have the Dec'07(rev1723) version of FFDShow installed.

To transcode aac -> ac3 files, use Foobar2000, BeHappy or Faad-Aften (command line). The GraphEdit-DirectShow filters method is dificult to support because each machine have very different DirectShow configuration.

The ffdshow method to transcode aac -> ac3 on the fly can't save CPU usage because not only need decode the aac but also encode to ac3 with more CPU usage.