Convert aac to wav to ??? to ac3

[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]

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? :confused: 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. :angry:

[johnman]

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


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]

:stupid: :p 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 (http://www.rarewares.org/aac.html)

Easy but long way without GraphEdit:

With aacDecDrop (http://www.rarewares.org/aac.html) (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]

:stupid:

thx :angry:

[magicclue]

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

:thanks: for the guides.

[johnman]

DOH..... clicked wrong face, what i meant was

thx :p

cant see how i missed that one

[Sakuya]

Thanks magicclue! :goodpost: I can't wait to try that. For AACDecDrop, it gives me an error message because it cannot find the dynamic link library libmmd.dll. :eek:

And to answer johnman's question, Hexedit crashes when I try to open the 6ch WAV. Winhex won't let me save anything above 200KB because it's the trial version.

[magicclue]

just download the libmmd.dll from www.rarewares.org as well...
put it in the according application folder
or -> in system32

you'll need that dll for alot of programs..

[Sakuya]

Thanks. It's working now. Just a thought. In BeSweet, it says this:

+------- BeSweet -----
| Input : file.wav
| Output: FL, FR, SL, SR, C, LFE
| Floating-Point Process: No
| Source Sample-Rate: 22.1KHz
+---------------------

Does that have to be 48000? Do I need to convert it to be 48000?

[johnman]

Thanks. It's working now. Just a thought. In BeSweet, it says this:

+------- BeSweet -----
| Input : file.wav
| Output: FL, FR, SL, SR, C, LFE
| Floating-Point Process: No
| Source Sample-Rate: 22.1KHz
+---------------------

Does that have to be 48000? Do I need to convert it to be 48000?

As you have noticed im no aac guru, but 22.1khz doesnt seem good to me. 22 khz smplrate means there is no sound above +- 11khz. And if you wonder about it, upsampling doesnt help once the samplerate has been lowerd.

[magicclue]

T... Does that have to be 48000? Do I need to convert it to be 48000?

? I don't know what you want to do or what type of source you use?!
If your source is bad it won't help.
Just leave as it is although I cannot think of why someone would record at that frequency :eek:

The only thing I can think of to convert to 48KHz would be to use that sound for a DVD.

[Sakuya]

Yes, I'm making a DVD. In Soft Encode, under Encode Settings, I set it to 48KHz and the end result is very fast-paced as if it was being fast-forwarded. :eek: So now I'm setting that option to auto, which is 44.1KHz and re-encoding. :( How can I get it to be 48KHz without it being fast-paced?

[johnman]

If it plays to fast it could mean the samplerate is wrong. If you i.e. play a 22khz at 44kz you also hear its to fast and high.
To change this you must not resample, but edit the samplerate to the correct value. Cool edit has an option for this called "Adjust samplerate". Maybe other porgrams can do the same thing but i dont know any(maybe besweet?) . Can you give the ratio by which it has been speed up. You can calculate this by dividing the short length by the original length.

[Sakuya]

The original length is 1 hour, 39 minutes, 45 seconds. The AC3 is 49 minutes, 53 seconds. Cool Edit Pro cannot load AC3 files. I tried AC3Machine but the result is the same fast and high-pitched. :(

[johnman]

The ratio between in and output is almost exactly 2, so my gues is this is caused playing 22 @ 44 khz.

Since you are using soft encode to encode it, im asuming you've got the sound as a wav. What you need to do is to make a good wav and encode it. So the first thing you should do is check if the wav is correct so first check this. If it has the correct length and plays correct you should upsample it to 48khz. Then you can just insert in surcode and let sonic SE encode it at 48khz. Then the ac3 should come out good.

If the original wav isnt correct you should deal with that first.

[Sakuya]

Yes, I checked and the 6ch WAV has a length of 1 hour 39 minutes and 45 seconds (or 99 minutes and 45 seconds). :( So the problem is with Soft Encode?

[johnman]

Yes, I checked and the 6ch WAV has a length of 1 hour 39 minutes and 45 seconds (or 99 minutes and 45 seconds). :( So the problem is with Soft Encode?

Is the original also 48 khz? If not upsample it first and THEN insert it in sonic SE.

[Sakuya]

How do I check if it is? :confused: And what do I use to upsample?

Edit:
I used SSRC to try to convert the sampling rate and it says:

22050 > 48000

Which means that it is 22050KHz. Now then, what do I use to upsample?

[johnman]

You can use ssrc to upsample, or wavewizard if you dont like cli's.
Wavewizard also shows the original samplerate.
For wavewizard look @ http://forum.doom9.org/showthread.php?t=95265

Remember to set channelmapping off in the preferences.

EDIT
ssrc can upsample, and you should actually use ssrc_hp

i believe you should do something like ssrc_hp --rate 48000 in.wav out.wav

[Sakuya]

I have tried ssrc_hp but it resulted in an erroneous WAV that won't play in Media Player Classic. Not to mention the file was extended to 2 hours and 4 minutes.

I have also tried WaveWizard but the result was "damaged" as the error in Media Player Classic said. So now, I will have to wait for the pictures to show up in the other method that magicclue posted. Meanwhile, any other ideas? :confused:

Edit:
I tried the ffdshow method and still the same problem with the sampling rate! Tomorrow when I have more time, I will open the 6ch WAV with Cool Edit and convert the sampling rate (adjusting the rate won't do any good, I've tried).

[johnman]

If you want to test the 6 ch stream you can make monofiles from them and check these with a program like CE. This way you can see what went wrong.
Wavewizard can also make monstreams from files.


I dont know mediaplayer classic, but i wouldnt use it for testing.

(maybe im lucky, but i havent had a bad stream from wavewizard nor ssrc ever)

EDIT and another way to test the file pops into my head :). Just drop the wrong file again in wavewizard. Wavewiz is pretty strict about the format so if ww accepts it, i cant immagina what's wrong.

EDIT 2 how big is the file?

[magicclue]

... I set it to 48KHz and the end result is very fast-paced as if it was being fast-forwarded. :eek: So now I'm setting that option to auto, which is 44.1KHz and re-encoding. :( How can I get it to be 48KHz without it being fast-paced?

Please please I BEG YOU
:readfaq:
:search:

There are alot of guides all over the froum and on the front page.
Read some basics and you'll be able to do all what you want!

A hint: ->besweet->ssrc

did you READ my conversion guide???
Well it's all there.

[tebasuna51]

1) Recent alpha versions of ffdshow can transcode directly from aac to ac3 (with GraphEdit). Is the best way ... but not for me, because I have others troubles with this versions and I return to the last "official" alpha 20041012. Maybe in the future.

2) If you need edit or resample the wav (Can ffdshow do the upsampling 44.1 KHz -> 48 KHz. on-the-fly?) you must use the long method.
To decode the aac I use Foobar2000 without problems. I tried now AacDrop and run Ok for me (only the downsampling to half frecuency when the aac is low quality)

A words to explain my method in previous posts. There are two limits for the size of the wavs (with canonical header) 2 and 4 GB. In the header there are a field of four bytes to put the length of data, if the soft read this four bytes like signed integer only can manage 2 GB, read like unsigned integer can manage 4 GB.
And more..., Faad, FooBar2000 and AacDrop can write corrects wav's greater than 4 GB with only an error in the field mentioned (and in a second field for the length of file).
BeSplit is the only soft, I know, to demux in 6 mono wavs this wav6 > 4 GB and after be open in a wav editor or SoftEncode.

Is interesting to know this limits in function of the time of the aac. For an aac 48 KHz:
Time.aac Size wav6 Header SoftEncode BeSweet WaveWizard BeSplit
________ ________ _____ __________ _______ __________ ______
_62m.aac _1.99 GB_ __OK_ ____OK___ __OK___ ____OK____ __OK
_63m.aac _2.03 GB_ __OK_ ____NO___ __NO___ ____OK____ __OK
124m.aac _3.99 GB_ __OK_ ____NO___ __NO___ ____OK____ __OK
125m.aac _4.02 GB_ __NO_ ____NO___ __NO___ ____NO____ __OK

(Limits for 44.1 KHz: 67/68 m and 135/136 m.)

Only for aac > 124 m. is necessary to use BeSplit and WinHex to correct a bug in the field BlockAlign (bug reported, but I think DspGuru is out).
Of course you can split the aac in parts, reencode to ac3, and after join them.

Can WaveWizard ignore the data length field, like BeSplit, and prompt if you want to split in mono wav's using the real length of the file?. Then the problem is resolved.

[johnman]

This is slightly OT but may be usefull.

Can WaveWizard ignore the data length field, like BeSplit, and prompt if you want to split in mono wav's using the real length of the file?

I just added this feature. The new release wil have an option in the preferences to ignore invalid wav size . This also implies the current version cant cant do this. Its also possible to use the wavewiz format as an intermediate format. This has a 64 bit number for the filesize (filesize can be 2^64).

[magicclue]

1) Recent alpha versions of ffdshow can transcode directly from aac to ac3 (with GraphEdit).

yep. see my guide above.

Can ffdshow do the upsampling 44.1 KHz -> 48 KHz. on-the-fly?
yes, no problem.


BeSplit is the only soft, I know, to demux in 6 mono wavs this wav6 > 4 GB
have you tried "Transcode" to separate the 6 channel wav to mono wavs?
It's a new Soft floating in the forum. It's just some weeks old.

Link to Transcode (http://forum.doom9.org/showthread.php?p=648990#post648990)

[Sakuya]

I think I've found my problem. I just realized that my AAC has a length of 1:39:45 while my AVI is 1:39:52. Stupid VFR MKV. I'll have to go to the other thread about this before proceeding with this audio.

Thanks for the help! :scared:

[tebasuna51]

Tranzcode works!

I have a 130 m. wav6 and open in tranzcode_GUI inform:
Duration: 7456.540 sec. (124.27 min. just 4 GB)
But after 30 min. waiting I have 6 mono wav's with the exact 130 min.

I tried it last week, but in the input dialog offers DTS and DTSWAV only and I have closed the program.
It work also for PCM WAV.

Thanks.

[magicclue]

Tranzcode works!
...It work also for PCM WAV...

wonderful!

yep. Kurtnoise should update the input dialog.
I'll mail him.

[Steffi]

hi!

Tried your way to convert aac to ac3, but my filters wont connect with fddshow? Any clue, why?
thx.

[Steffi]

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"

[see figure 1]

-swap channels->choose according channel mapping exactly as seen in figure 2!
-click ok

[see figure 2]

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

[see figure 3]


my filters wont connect, any clue?

[tebasuna51]

In http://forum.doom9.org/showthread.php?t=98133&page=2 I wrote:

"@magicclue
I report two differences with your method aac -> ac3 with ffdshow:

1) In GraphEdit I can't connect directly any aac file with ffdshow audio decoder. With Connect Intelligent disabled send me a error, enabled insert a aac_parser: AAC_parser from http://www.rarewares.org/aac.html or nero_aac_parser.

2) I don't need to remap any channel. With channelmapping disabled I make a correct ac3."

For me work with: aac -> AAC_Parser -> ffdshow -> file writer
And without channelmapping.

[magicclue]

@tebasuna51: thanks. Guide corrected.

As for filters in GraphEdit not connecting to ffshow audio decoder: activate "allow ffdshow to connect to any filter" and "direct-file-output".
AND activate FFDSHOW as decoder for the file format you want to convert (see codec tab)!
So activate ffdhsow for aac/m4a
Else I don't know.

[tebasuna51]

I have:
Windows XP Pro, SP 1
GraphEdit (Build 040927) from http://www.3ivx.com/download/windows.html
ffdshow 2005-08-03 with:

Codecs -> AAC - realaac (Enabled)
Connect to: -> any filter
Allow direct-to-file output: Enabled
Don't use Waveformatextensible header when not needed: Enabled

And still need a AAC_parser.
No problem for me (works fine) but maybe the Steffi's problem.

[Rockaria]

I have the same configuration except the graphedit(build 041201). But the ffdshow didn't need the AAC_Parser for both AAC and MP4. AC3Filter also only needed the CoreAAC directshow filter.

AAC parser filter for DirectShow 2003-11-25
Parser for decoding unwrapped AAC streams in DirectShow compatible players (Windows Media Player, Media Player Classic, RadLight, ZoomPlayer...) - requires an AAC decoder filter (CoreAAC or 3ivX's) - by Tom Judd - v 1.1.0

Maybe the AAC formats are different? Which AAC encoder did you use? I used nero he+AAC encoder to make the the source in the foobar2k.

[tebasuna51]

@Rockaria
The same wav 6 channel converted with different method (and extension):

Belight_Nero_Plugin.mp4: directly to ffdshow
Faac1.24+/Foobar_faac.mp4: directly to ffdshow

Faac1.24+/Foobar_faac.aac: need AAC_Parser
WaveWizard_NeroFends_ISO_13818-7.aac: need AAC_Parser
Ahead AAC Encoder v2.9_Frontend v0.74.aac: need AAC_Parser
PsyTEL(R) MPEG-4 AAC Encoder V2.15.aac: need AAC_Parser
Track2.aac from Matrix Reloaded Trailer-HE_AAC_sub.mkv: need AAC_parser

Faac1.24+/Foobar_faac.m4a: need 3ivx D4 Media Splitter
WaveWizard_NeroFends_NO_ISO_13818-7.aac: need 3ivx D4 Media Splitter

[Rockaria]

9. What is the difference between *.AAC and *.MP4 and *.M4A?

AAC files usually contain AAC with ADTS headers or raw AAC data streams. Raw AAC cannot be processed without decoding some of the stream. MP4, on the other hand, is a container format of the MPEG4 standard which can contain AAC streams and many other things. The data format of the two is very different and hence you cannot rename them to one another, they must be muxed (put into the container) or demuxed. Muxing of AAC into MP4 and demuxing can be done with Ivan & Menno. M4A is normally audio in an MP4 container that has been renamed to clarify that it is audio only. Itunes, Winamp5, and Realplayer gold encode to M4A now by default. Note that M4A and MP4 files can be switched to each other by simply renaming, since they are both extensions for the same container format.

Note that the "Export ISO 13818-7 Stream" option in the Nero encoder produces *.AAC files with ADTS headers. Again this is not a raw aac stream, Ivan & Menno is required to convert to raw aac.

7. What do I need for AAC playback?

Download and install the CoreAAC DirectShow filter or the 3ivx audio decoder (bundled with 3ivx codec), both are excellent for AAC (Both LC and HE) playback in any directshow compatible media player. For the latest CoreAAC filter and winamp/foobar plugins, refer to john33's thread at HydrogenAudio.org.

Note that for MP4 streams, 3ivx or another MPEG-4 splitter is required for playback in DirectShow compatible media players. For raw AAC streams, the AACparser filter from rarewares is required. When muxing in OGM or Matroska, an appropriate splitter is needed as well. See the FAQs in the New Container Formats forum for more information.


Quoted from the Audio FAQ (http://forum.doom9.org/showthread.php?p=424070#post424070) sticky.

[tebasuna51]

@Rockaria
Yes. I know that.

Faac1.24+.m4a and Faac1.24+.mp4 are identical internally (and Foobar.m4a with Foobar.mp4). Then, why don't can connect, in GraphEdit, with ffdshow the .m4a like the .mp4?

The answer can't be the internal format (there are identical), I think GraphEdit works different in each system (installed DirectShow filters, ID, merit, ...), and for that my post.

[Rockaria]

@Rockaria
Yes. I know that.
Yeah, you seem to be knowing it already. :stupid:

Actually my posts were for anybody who has trouble in connecting the ffdshow dsfilter to an uncertain format AAC file.
And nobody knows better than oneself about the own system, now very few things to say in common....

[Lothar]

After using faad (else i get a 8-ch wave or a downsampled one) to convert from aac to 6ch-wave, I need to change the duration to obtain a 25-frames ac3 pal from a 23.9376-frames aac ntsc, and even after editing the header with winhex (or using wavewizard) besweet doesn't work with the 6ch-wave.

The only method that gets me to have the 6 working waves is using tranzcode to have the 6 waves (correct order, it seams).

But I can't create a mono-wave from a mono-wave using besweet....

Thanks for the help!

[tebasuna51]

1) You don't need chage the duration of audio, you need change the framerate of the video from 23.9376 to 25 fps preserving the duration. Virtualdub can insert a DUP frame to obtain a new 25 fps video with more frames but with the same duration:
Virtualdub -> Video -> Direct stream copy
Virtualdub -> Video -> Framerate -> Frame rate converssion -> Convert to fps: 25

2) If you use:
Virtualdub -> Video -> Framerate -> Source rate adjustment -> Change to 25 fps
Then VirtualDub don't insert DUP frames and the video play more quickly than real movie. In this case you need a more short audio. Use Audacity (any audio editor) over the 6 mono wav, or if you want use BeSweet, you can merge the 6 mono wav in 3 stereo wav with WaveWizard or with a .mux file with BeSweet

I always use the first method without change the audio.

[Lothar]

I used audition instead of audacity (i don't know why but with audacity i couldn't get the right duration) to adjust the speed of the mono waves, and it seems everything is fine now.

Thx for the help.

[blizzard]

I used the graphedit way to convert the he-aac to ac3 and it does work but there is something i don't understand.
For example, if the original audio is 160kb/s while he-aac which compression should i choose in ffdshow so that it will lose the minimum quality possible?
If i have 160kb/s at start and in ffdshow choose 160kb/s the resulting sound will be crappy...so it should need an higher number!
Is there any kind of ratio that i should be aware?
Thank you

[Rockaria]

If we suppose the 160kbps he-aac contains 80% of the quality, you cannot achieve even the same 80% fidelity with the lossy AC3 codec at it's full 640kbps bandwidth.
So minimizing the transcoding chain is always advised in lossy encoding.

[tebasuna51]

@blizzard
If you want convert to ac3 I think the aac is a multichannel 5.1. Then 160 Kbps for a 5.1 is a low quality, but if you don't want lose more quality you must be generous with the ac3 bitrate.

A parameter to be considered is the bandwidth:
A 5.1 ac3 with 224 kbps have 9.05 KHz , 256 -> 12.42, 320 -> 15.8, 384 -> 18.05, 448 -> 20.3 KHz (maximum).
By my test I think that HE-aac do the first 9 KHz like LC and the SBR section do the band 9-18KHz (with less quality).
Then, if you don't want lose nothing, you need 384 kbps (bandwith 18.05 KHz), smaller bitrate lose high frequencies.

[Rockaria]

The bandwidth by tebasuna51 is pricisely the cutoff frequency(tone) of the sound while mine is the bitrate per second to sample and compress the wav(pcm) signal to ac3.
The codecs tend to cut off the high frequencies(inaudible area) to achieve the pleasing sound for human with less size(bit rate), which affect the overall quality very little.
Over 17khz is almost audiophile zone : supersonic.

If you look at the ffdshow-> output, you will find the ac3 bit rate upto 640.
What I can suggest is to keep the original format from the aac :
. sample size(probably 16) : the dynamic volume range also affect the quality
. sample rate(mostly 44.1khz or 48khz) : 48khz is almost a standard to pc or external devices

Just change the bit rate between 192~640 to get the proper target quality.
You might also want to control the volume to get proper gain(volume + DRC)

What is better with the ffdshow method is you can listen what you get by attaching the default sound device before doing the main job.